Commit Graph

23356 Commits

Author SHA1 Message Date
Jeeja KP 84c9e2836a ASoC: Intel: Skylake: Correct the runtime handler behaviour
On runtime pm resume, we need to download the firmware, also on
suspend we need to ensure all the interrupts from controller and
DSP are disabled.

Also since we download the firmware on resume, we don't need to do
so on init, so remove that bit

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-09 11:19:00 +01:00
Jeeja KP def656fe22 ASoC: Intel: Skylake: Verify the status bit before handling interrupt
Like we have in legacy mode HDA driver, we need to check the
status bit and handle interrupt only when it is not zero or all
bits set.  We typically see the status as all 1's when controller
resumes from suspend, So add the check here as well and don't
handle for these cases.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-09 11:18:59 +01:00
Jeeja KP 6ea8ba33e6 ASoC: Intel: Skylake: Check CPA bit in DSP core power down
Skylake driver will set the SPA bit to 0 to turn off the DSP core.
Driver will poll the Current Power Active (CPA) bit to match the
Set Power Active (SPA) bit value. When CPA bit matches the value
of SPA bit, the achieved power state has reached.

In case of DSP power down, register that was polled is SPA
instead of CPA. This patch corrects the register to be polled
in case of DSP power down.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-09 11:18:58 +01:00
Takashi Sakamoto bde3e2880f ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error
Currently, when asynchronous transactions finish in error state and
retries, work scheduling and work running also continues. This
should be canceled at fatal error because it can cause endless loop.

This commit enables to cancel transferring MIDI messages when transactions
encounter fatal errors. This is achieved by setting error state.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:06 +02:00
Takashi Sakamoto ea848b7b62 ALSA: firewire-lib: add throttle for MIDI data rate
Typically, the target devices have internal buffer to adjust output of
received MIDI messages for MIDI serial bus, while the capacity of the
buffer is limited. IEEE 1394 transactions can transfer more MIDI messages
than MIDI serial bus can. This can cause buffer over flow in device side.

This commit adds throttle to limit MIDI data rate by counting intervals
between two MIDI messages. Usual MIDI messages consists of two or three
bytes. This requires 1.302 to 1.953 mili-seconds interval between these
messages. This commit uses kernel monotonic time service to calculate the
time of next transaction.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:06 +02:00
Takashi Sakamoto e8a40d9bcb ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages
Currently, when two MIDI trigger callbacks can be called immediately,
transactions for the second MIDI messages can be postpone till next trigger
callback. This is not good for real-time message transmission.

This commit schedules work again at response handling callback if the
MIDI substream still includes untransferred MIDI messages.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:05 +02:00
Takashi Sakamoto d3ef9cb93a ALSA: firewire-lib: add a restriction for a transaction at once
Currently, when waiting for a response, callers can start another
transaction by scheduling another work. This is not good for error
processing of transaction, especially the first response is too late.

This commit serialize request/response transactions, by adding one
boolean member to represent idling state.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:05 +02:00
Takashi Sakamoto 585d7cba5e ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages
Some models receive MIDI messages via IEEE 1394 asynchronous transactions.
In this case, MIDI messages are transferred in fixed-length payload. It's
nice that firewire-lib module has common helper functions.

This commit implements this idea. Each driver adds
'struct snd_fw_async_midi_port' in its instance structure. In probing,
it should call snd_fw_async_midi_port_init() to initialize the
structure with some parameters such as target address, the length
of payload in a transaction and a pointer for callback function
to fill the payload buffer. At 'struct snd_rawmidi_ops.trigger()'
callback, it should call 'snd_fw_async_midi_port_run()' to start
transactions. Each driver should ensure that the lifetime of MIDI
substream continues till calling 'snd_fw_async_midi_port_finish()'.

The helper functions support retries to transferring MIDI messages when
transmission errors occur. When transactions are successful, the helper
functions call 'snd_rawmidi_transmit_ack()' internally to consume MIDI
bytes in the buffer. Therefore, Each driver is expected to use
'snd_rawmidi_transmit_peek()' to tell the number of bytes to transfer to
return value of 'fill' callback.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:57:04 +02:00
Kosuke Tatsukawa 694470273d ALSA: seq_oss: fix waitqueue_active without memory barrier in snd-seq-oss
snd_seq_oss_readq_put_event() seems to be missing a memory barrier which
might cause the waker to not notice the waiter and miss sending a
wake_up as in the following figure.

    snd_seq_oss_readq_put_event		    snd_seq_oss_readq_wait
------------------------------------------------------------------------
					/* wait_event_interruptible_timeout */
					 /* __wait_event_interruptible_timeout */
					  /* ___wait_event */
					  for (;;) {									 prepare_to_wait_event(&wq, &__wait,
					    state);
spin_lock_irqsave(&q->lock, flags);
if (waitqueue_active(&q->midi_sleep))
/* The CPU might reorder the test for
   the waitqueue up here, before
   prior writes complete */
					  if ((q->qlen>0 || q->head==q->tail)
					  ...
					  __ret = schedule_timeout(__ret)
if (q->qlen >= q->maxlen - 1) {
memcpy(&q->q[q->tail], ev, sizeof(*ev));
q->tail = (q->tail + 1) % q->maxlen;
q->qlen++;
------------------------------------------------------------------------

There are two other place in sound/core/seq/oss/ which have similar
code.  The attached patch removes the call to waitqueue_active() leaving
just wake_up() behind.  This fixes the problem because the call to
spin_lock_irqsave() in wake_up() will be an ACQUIRE operation.

I found this issue when I was looking through the linux source code
for places calling waitqueue_active() before wake_up*(), but without
preceding memory barriers, after sending a patch to fix a similar
issue in drivers/tty/n_tty.c  (Details about the original issue can be
found here: https://lkml.org/lkml/2015/9/28/849).

Signed-off-by: Kosuke Tatsukawa <tatsu@ab.jp.nec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-09 09:45:52 +02:00
Vinod Koul 70b4891cc8 ALSA: hda: make use of core codec fns
Now that we have introduced the core fns we should make hda use these
helpers

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-08 19:09:36 +02:00
Subhransu S. Prusty 1b5e6167c2 ALSA: hdac: Copy codec helpers to core
The current codec helpers are local to hda code and needs to be moved to
core so that other users can use it.
The helpers to read/write the codec and to check the
power state of widgets is copied

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-08 19:09:30 +02:00
Sudip Mukherjee dc542fb417 ASoC: rt5645: fix build warning
We were getting build warning about "Section mismatch".
dmi_platform_intel_broadwell is being referenced from the probe function
rt5645_i2c_probe(), but dmi_platform_intel_broadwell was marked with
__initdata.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Reviewed-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-08 16:15:00 +01:00
Sjoerd Simons f874b80e15 ASoC: rockchip: Add rockchip SPDIF transceiver driver
Add a driver for the SPDIF transceiver available on RK3066, RK3188 and
RK3288. Heavily based on the rockchip i2s driver.

Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-08 16:12:01 +01:00
Fang, Yang A b3681308cc ASoC: nau8825: add acpi match ID
This patch adds the acpi match ID for nau8825 codec

Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-08 14:38:42 +01:00
Vinod Koul 3373f71683 ASoC: Intel: Skylake: Modify the log level
dev_info is too noisy for tplg wiget loading, so move it to
debug level

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-08 09:46:22 +01:00
Takashi Iwai 601d62959d ASoC: Fixes for v4.3
Quite a few fixes here but they're all very small and driver specific,
 none of them really stand out if you aren't using the relevant hardware
 but they're all useful if you do happen to have an affected device.
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Merge tag 'asoc-fix-v4.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v4.3

Quite a few fixes here but they're all very small and driver specific,
none of them really stand out if you aren't using the relevant hardware
but they're all useful if you do happen to have an affected device.
2015-10-07 20:11:21 +02:00
Mark Brown e4fc141d2a Merge remote-tracking branches 'asoc/fix/tlv320aic3x' and 'asoc/fix/wm8962' into asoc-linus 2015-10-07 16:07:50 +01:00
Mark Brown 1e2fa4cfdb Merge remote-tracking branches 'asoc/fix/db1200', 'asoc/fix/dwc', 'asoc/fix/imx-ssi', 'asoc/fix/maintainers', 'asoc/fix/rt5645', 'asoc/fix/sgtl5000' and 'asoc/fix/tas2552' into asoc-linus 2015-10-07 16:07:16 +01:00
Jeeja KP 87b2bdf022 ASoC: Intel: Skylake: Initialize NHLT table
Load and Initialize Non HDA Link Table in Skylake driver
to get platform configuration.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 16:04:17 +01:00
Jeeja KP 2a29b200c6 ASoC: Intel: Skylake: Add DSP support and enable it
If processing pipe capability is supported, add DSP support.
Adds initialization/free/suspend/resume DSP functionality.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 16:04:16 +01:00
Jeeja KP b663a8c5c9 ASoC: Intel: Skylake: Initialize and load DSP controls
Initialize and creates DSP controls if processing pipe capability
is supported by HW. Updates the dma_id, hw_params to module param
to be used when DSP module has to be configured.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 16:04:15 +01:00
Vinod Koul 3af36706ff ASoC: Intel: Skylake: Add topology core init and handlers
The SKL driver does not code DSP topology in driver. It uses the
newly added ASoC topology core to parse the topology information
(controls, widgets and map) from topology binary.
Each topology element passed private data which contains
information that driver used to identify the module instance
within firmware and send IPCs for that module to DSP firmware
along with parameters.
This patch adds init routine to invoke topology load and callback
for topology creation.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 16:04:14 +01:00
Vinod Koul cfb0a87383 ASoC: Intel: Skylake: Add FE and BE hw_params handling
For FE and BE, the PCM parameters come from FE and BE hw_params
values passed. For a FE we convert the FE params to DSP expected
module format and pass to DSP. For a BE we need to find the
gateway settings (i2s/PDM) to be applied. These are queried from
NHLT table and applied.

Further for BE based on direction the settings are applied as
either source or destination parameters.

These helpers here allow the format to be calculated and queried
as per firmware format.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 15:30:16 +01:00
Vinod Koul d93f8e550f ASoC: Intel: Skylake: add DSP platform widget event handlers
The Skylake driver topology model tries to model the firmware
rule for pipeline and module creation.
The creation rule is:
 - Create Pipe
 - Add modules to Pipe
 - Connect the modules (bind)
 - Start the pipes

Similarly destroy rule is:
 - Stop the pipe
 - Disconnect it (unbind)
 - Delete the pipe

In driver we use Mixer, as there will always be ONE mixer in a
pipeline to model a pipe. The modules in pipe are modelled as PGA
widgets.  The DAPM sequencing rules (mixer and then PGA) are used
to create the sequence DSP expects as depicted above, and then
widget handlers for PMU and PMD events help in that.

This patch adds widget event handlers for PRE/POST PMU and
PRE/POST PMD event for mixer and pga modules.  These event
handlers invoke pipeline creation, destroy, module creation,
module bind, unbind and pipeline bind unbind

Event handler sequencing is implement to target the DSP FW
sequence expectations to enable path from source to sink pipe for
Playback/Capture.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 15:30:15 +01:00
Jeeja KP f7590d4f15 ASoC: Intel: Skylake: Add module configuration helpers
To configure a module, driver needs to send input and output PCM
params for a module in DSP. The FE PCM params come from hw_params
ie from user, for a BE they also come from hw_params but from
BE-link fixups.
So based on PCM params required driver has to find a converter
module (src/updown/format) and then do the conversion and
calculate PCM params in these pipelines In this patch we add the
helper modules which allow driver to do these calculations.

Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 15:30:15 +01:00
Jeeja KP e4e2d2f452 ASoC: Intel: Skylake: Add pipe and modules handlers
SKL driver needs to instantiate pipelines and modules in the DSP.
The topology in the DSP is modelled as DAPM graph with a PGA
representing a module instance and mixer representing a pipeline
for a group of modules along with the mixer itself.

Here we start adding building block for handling these. We add
resource checks (memory/compute) for pipelines, find the modules
in a pipeline, init modules in a pipe and lastly bind/unbind
modules in a pipe These will be used by pipe event handlers in
subsequent patches

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 15:30:15 +01:00
Adam Thomson 6e7c444318 ASoC: da7213: Add support to handle mclk data provided to driver
Driver now can make use of mclk data, if provided, to set, enable
and disable the clock source. As part of this, the choice to
enable clock squaring is dealt with as part of dai_sysclk() call
rather than as platform data.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 15:11:34 +01:00
Adam Thomson e90996a3ea ASoC: da7213: Add DT support to codec driver
This patch adds support for DT bindings in the codec driver.
As part of this support, the mclk data can now be provided and
used to control the mclk during codec operation.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 15:11:34 +01:00
Daniel Vetter 2844659842 Merge remote-tracking branch 'takashi/topic/drm-sync-audio-rate' into drm-intel-next-queued
Pull in the i915/hda changes for N/CTS setting so I can apply the
follow-up documentation work for drm/i915.

Some conflicts because ofc we had to rework i915 while that N/CTS work
was going on. But not more than adjacent changes really.

Signed-off-by: Daniel Vetter <daniel.vetter@intel.com>
2015-10-07 16:05:04 +02:00
Geert Uytterhoeven dcc448e619 ASoC: rsnd: Remove obsolete platform data support
Since commit 3d7608e4c1 ("ARM: shmobile: bockw: remove legacy
board file and config"), Renesas R-Car SoCs are only supported in
generic DT-only ARM multi-platform builds.  The driver doesn't need to
use platform data anymore, hence remove platform data configuration.

Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed
by platform code.

Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 12:19:51 +01:00
Adam Thomson ba856fbd60 ASoC: da7219: Improve error checking of mclk enable/disable
Should only try to enable/disable the provided mclk, during bias
level changes, if it's not NULL. Also return value of
clk_prepare_enable() should be checked and dealt with accordingly.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 12:15:22 +01:00
Adam Thomson b7ebd78d1d ASoC: da7219: Use of_match_ptr() when assigning match table
Use of_match_ptr() to handle non-DT kernel scenario where match
table should be NULL.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 12:15:22 +01:00
Maxime Ripard c570b82c5e ASoC: sun4i-codec: Remove the routing property
Most of the boards have their headphone jack directly connected to the
matching pins of the SoCs. Since most of the time we will have the same
routing path, it makes no sense to put that in the DTS, since it will only
be some useless duplication there.

It also fixes the following warning messages that were seen so far, on
boards where we were using the bindings in the documentation example.

sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack

Reported-by: Priit Laes <plaes@plaes.org>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 12:13:42 +01:00
kbuild test robot cc91ef0fd4 ASoC: nau8825: fix platform_no_drv_owner.cocci warnings
sound/soc/codecs/nau8825.c:1096:3-8: No need to set .owner here. The core will do it.

 Remove .owner field if calls are used which set it automatically

Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci

CC: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 11:36:39 +01:00
Anatol Pomozov 34ca27f34f ASoC: nau8825: Add driver for headset chip Nuvoton 8825
Sponsored-by: Google Chromium project
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 11:36:38 +01:00
Andy Shevchenko b5e5a4549c ASoC: Intel: use dw_dmac autoconfiguration
Instead of hardconding a platform data for dw_dmac let's use it's own
autoconfiguration feature. Thus, remove hardcoded values.

Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 11:24:49 +01:00
Axel Lin dc6d84c69c ASoC: rt286: Fix run time error while modifying const data
Make a copy of memory for index_cache rather than directly use the
rt286_index_def to avoid run time error.

Fixes: c418a84a8c ("ASoC: Constify reg_default tables")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 10:46:49 +01:00
Axel Lin d46183efe3 ASoC: ad193x-spi: Fixup ad193x_spi_id table
AD1939 is missed from the table, so add it.
AD1936 and AD1937 are controlled by I2C interface, so remove them.

Fixes: e5224f58e3 ("ASoC: ad193x: add support to ad1934")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-07 10:42:54 +01:00
Andreas Dannenberg e2600460bc ASoC: tas2552: fix dBscale-min declaration
The minimum volume level for the TAS2552 (control register value 0x00)
is -7dB however the driver declares it as -0.07dB.

Running amixer before the patch reports:
dBscale-min=-0.07dB,step=1.00dB,mute=0

Running amixer with the patch applied reports:
dBscale-min=-7.00dB,step=1.00dB,mute=0

Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
2015-10-06 11:53:46 +01:00
Sjoerd Simons 698d0b59f3 ASoC: rockchip: namespace rockchip i2s module name
Change the rockchip i2s object name (and thus module name) from the
rather generic snd-soc-i2s to the more specific snd-soc-rockchip-i2s

Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-06 11:52:47 +01:00
Cyrille Pitchen e5224f58e3 ASoC: ad193x: add support to ad1934
The AD1934 codec has no ADC feature. Hence it register mapping is slightly
different from the register mapping of other members of the AD193x family.

Some ASoC controls and widgets are related to the DAC feature so are not
relevant in the case of an AD1934 codec.

Signed-off-by: Cyrille Pitchen <cyrille.pitchen@atmel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 17:42:49 +01:00
Zidan Wang c64c60763b ASoC: fsl_esai: Add driver suspend and resume to support MEGA Fast
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, ESAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 17:26:19 +01:00
Zidan Wang 05cf237972 ASoC: fsl_ssi: Add driver suspend and resume to support MEGA Fast
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power
source of modules during system suspend and resume procedure. Thus,
SSI needs to save all the values of registers before the system
suspend and restore them after the system resume.

The register SFCSR is volatile, but some bits in it need to be
recovered after suspend/resume.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 17:26:18 +01:00
Zidan Wang f9f4fa61aa ASoC: fsl_spdif: Add driver suspend and resume to support MEGA Fast
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SPDIF needs to save
all the values of registers before the system suspend and restore them after
the system resume.

The SRPC register should be volatile, LOCK bit is set by the hardware.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 17:26:17 +01:00
Zidan Wang 1fde5e83a1 ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 17:26:16 +01:00
Zidan Wang 43ac946922 ASoC: imx-spdif: add snd_soc_pm_ops for spdif machine driver
Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume
be called in suspend/resume.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 17:25:45 +01:00
Maruthi Srinivas Bayyavarapu 1d957d862a ASoC: dwc: support dw i2s in slave mode
dw i2s controller can work in slave mode, codec being master.
dw i2s is made to support master/slave operation, by reading dwc
register.

Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 16:31:43 +01:00
Jeeja KP a04267fd87 ALSA: hdac: Fix to check if stream not in use in release
if the stream is decoupled and both link and host are used, while
releasing the stream, need to check if link and host stream are
not in use. This patch adds fix to check if the host/link stream
is in used before coupling it back when releasing the stream.

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-05 17:18:56 +02:00
Subhransu S. Prusty 88b19968a2 ALSA: hdac: Fix incorrect update of stream id mapping
Bits in LOSIDV need to be set to map the stream id for specific link.
Fixing this by setting the required bits in the register.

Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-05 17:18:44 +02:00
Adam Thomson 4e929134eb ASoC: da7219: Improve error handling for regulator supplies
Currently if bulk_enable() of supplies fails, the code still goes on
to try and put the device into active state, and set expected IO
voltage of the device. This doesn't really make sense so code now
returns on bulk_enable() failure, with an error message. Also,
to help with debug, failure to get supplies also provides an error
message.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 15:59:16 +01:00
Takashi Iwai 225db5762d ALSA: synth: Fix conflicting OSS device registration on AWE32
When OSS emulation is loaded on ISA SB AWE32 chip, we get now kernel
warnings like:
  WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80()
  sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0'

It's because both emux synth and opl3 drivers try to register their
OSS device object with the same static index number 0.  This hasn't
been a big problem until the recent rewrite of device management code
(that exposes sysfs at the same time), but it's been an obvious bug.

This patch works around it just by using a different index number of
emux synth object.  There can be a more elegant way to fix, but it's
enough for now, as this code won't be touched so often, in anyway.

Reported-and-tested-by: Michael Shell <list1@michaelshell.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-05 16:55:09 +02:00
Oder Chiou a2c026cfec ASoC: rt5645: Prevent the weird sound of the headphone while rebooting
The patch adds the codec reset setting in the shutdown function to prevent
the weird sound of the headphone happened by rebooting.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-05 15:27:16 +01:00
Takashi Iwai c7e1008048 ALSA: hda - Disable power_save_node for IDT 92HD73xx chips
The recent widget power saving introduced some unavoidable click
noises on old IDT 92HD73xx chips while it still seems working on the
compatible new chips.  In the bugzilla, we tried lots of tests and
workarounds, but they didn't help much.  So, let's disable the feature
for these specific chips as the least (but safest) fix.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-04 22:47:40 +02:00
Takashi Sakamoto 425a570e1b ALSA: bebob: support Firewire I/O card of Mackie Onyx 1220/1620/1640
Current ALSA BeBoB drivers has an entry for this model, while the value of
vendor ID seems to be wrong according to an user's report.

The vendor had released no updated firmware, thus we can judge that this
model had not changed the content of its config ROM. It's reasonable to fix
the ID according to the report.

$ ./linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom

               ROM header and bus information block
               -----------------------------------------------------------------
400  0425720f  bus_info_length 4, crc_length 37, crc 29199
404  31333934  bus_name "1394"
408  f0646122  irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100,
               max_rec 6 (128), max_rom 1, gen 2, spd 2 (S400)
40c  00000ff2  company_id 00000f     |
410  00004697  device_id f200004697  | EUI-64 00000ff200004697

               root directory
               -----------------------------------------------------------------
414  000859be  directory_length 8, crc 22974
418  04000082  hardware version
41c  0c0083c0  node capabilities per IEEE 1394
420  03000ff2  vendor
424  8100000a  --> descriptor leaf at 44c
428  17010065  model
42c  8100000d  --> descriptor leaf at 460
430  13000910  version
434  d1000001  --> unit directory at 438

               unit directory at 438
               -----------------------------------------------------------------
438  0004ccec  directory_length 4, crc 52460
43c  1200a02d  specifier id: 1394 TA
440  13010001  version: AV/C
444  17010065  model
448  8100000d  --> descriptor leaf at 47c

               descriptor leaf at 44c
               -----------------------------------------------------------------
44c  0004152a  leaf_length 4, crc 5418
450  00000000  textual descriptor
454  00000000  minimal ASCII
458  4d61636b  "Mack"
45c  69650000  "ie"

               descriptor leaf at 460
               -----------------------------------------------------------------
460  000612b5  leaf_length 6, crc 4789
464  00000000  textual descriptor
468  00000000  minimal ASCII
46c  4f6e7978  "Onyx"
470  20466972  " Fir"
474  65776972  "ewir"
478  65000000  "e"

               descriptor leaf at 47c
               -----------------------------------------------------------------
47c  000612b5  leaf_length 6, crc 4789
480  00000000  textual descriptor
484  00000000  minimal ASCII
488  4f6e7978  "Onyx"
48c  20466972  " Fir"
490  65776972  "ewir"
494  65000000  "e"

$ cat /proc/asound/card3/firewire/firmware
Manufacturer:   bridgeCo
Protocol Ver:   1
Build Ver:      0
GUID:           0x00000FF200004697
Model ID:       0x82
Model Rev:      1
Firmware Date:  20040430
Firmware Time:  131527
Firmware ID:    0x10065
Firmware Ver:   2320
Base Addr:      0x20080000
Max Size:       1572864
Loader Date:    20040430
Loader Time:    112036

Reported-by: Andrzej Gansiniec <andrzej@gansiniec.pl>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-04 17:21:34 +02:00
Viresh Kumar 621a5f7ad9 debugfs: Pass bool pointer to debugfs_create_bool()
Its a bit odd that debugfs_create_bool() takes 'u32 *' as an argument,
when all it needs is a boolean pointer.

It would be better to update this API to make it accept 'bool *'
instead, as that will make it more consistent and often more convenient.
Over that bool takes just a byte.

That required updates to all user sites as well, in the same commit
updating the API. regmap core was also using
debugfs_{read|write}_file_bool(), directly and variable types were
updated for that to be bool as well.

Signed-off-by: Viresh Kumar <viresh.kumar@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
2015-10-04 11:36:07 +01:00
John Flatness e8ff581f7a ALSA: hda - Apply SPDIF pin ctl to MacBookPro 12,1
The MacBookPro 12,1 has the same setup as the 11 for controlling the
status of the optical audio light. Simply apply the existing workaround
to the subsystem ID for the 12,1.

[sorted the fixup entry by tiwai]

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105401
Signed-off-by: John Flatness <john@zerocrates.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-03 10:32:49 +02:00
Laura Abbott d05ea7da0e ALSA: hda: Add dock support for ThinkPad T550
Much like all the other Lenovo laptops, add a quirk to make
sound work with docking.

Reported-and-tested-by: lacknerflo@gmail.com
Signed-off-by: Laura Abbott <labbott@fedoraproject.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-03 10:31:08 +02:00
Ben Zhang fbe039bb08 ASoC: rt5645: Allow 4 channel recording on AIF1
The codec supports 4 channel recording with TDM on AIF1.
This patch modifies the DAI capability to allow it.

Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-02 18:13:06 +01:00
Adam Thomson 6d817c0e9f ASoC: codecs: Add da7219 codec driver
This adds support for the DA7219 audio codec with built-in advanced
accessory detect features.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-02 18:11:27 +01:00
yitian 924eb47512 ASoC: dwc: fix dma stop transferring issue
Designware I2S uses tx empty and rx available signals as the DMA
handshaking signals. during music playing, if XRUN occurs,
i2s_stop() function will be executed and both tx and rx irq are
masked, when music continues to be played, i2s_start() is executed
but both tx and rx irq are not unmasked which cause I2S stop
sending DMA handshaking signal to DMA controller, and it finally
causes music playing will be stopped once XRUN occurs for the first
time.

[On list discussion suggests this may be partly a race condition on slow
systems -- broonie]

Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-02 18:05:51 +01:00
Vedang Patel 5b2688a59a ASoC: topology: ABI - Add PCM Support and bump ABI version to 4
The struct snd_soc_tplg_pcm_dai is renamed to snd_soc_tplg_pcm.
This struct will now be used to handle data related to PCMs
(FE DAI & DAI links). It's not for BE, because BE DAI mappings will be
provided by ACPI/FDT data.

Remove the unused struct snd_soc_tplg_pcm_cfg_caps. We are using
snd_soc_tplg_stream and snd_soc_stream_caps instead.

Bump ABI version to 4.

Signed-off-by: Vedang Patel <vedang.patel@intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-02 17:59:48 +01:00
Richard Fitzgerald 6ebbce0a6e ASoC: wm8998: Initial WM8998 codec driver
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>
2015-10-02 17:51:34 +01:00
Richard Fitzgerald 50e6168f27 ASoC: arizona: add defines for single-input gain control
Adds convenience defines for declaring a gain control that
has an input mux. These blocks are functionally equivalent to
the existing mixer blocks but can only have a single input
active at once.

Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-02 17:51:33 +01:00
Takashi Sakamoto e5e0c3dd25 ALSA: firewire-tascam: add hwdep interface
This commit adds hwdep interface so as the other IEEE 1394 sound devices
has.

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>
2015-10-02 18:17:04 +02:00
Takashi Sakamoto e453df44f0 ALSA: firewire-tascam: 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 as an userspace application requests.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-02 18:17:03 +02:00
Takashi Sakamoto 35efa5c489 ALSA: firewire-tascam: add streaming functionality
This commit adds streaming functionality for both direction. To utilize
the sequence of the number of data blocks in packets, full duplex with
synchronization is applied.

Besides, TASCAM FireWire series allows drivers to decide which PCM data
channels are enabled. For convenience, this driver always enable whole the
data channels.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-02 18:17:02 +02:00
Takashi Sakamoto 47faeea25e ALSA: firewire-tascam: add data block processing layer
TASCAM FireWire series uses non-blocking transmission for AMDTP packet
streaming, while the format of data blocks is unique.

The CIP headers includes specific value in FMT field and no SYT
information.

In transmitted packets, the first data channel represents event counter,
and the last data channel has status and control information. The rest
has 24bit PCM samples with right padding.

In received packets, all of data channels include 16, 24, 32bit PCM
samples. There's no other kind of information.

This commit adds support for this protocol. For convenience, the size of
PCM samples in outgoing packet is limited by 16 and 24bit. The status and
control information will be supported in future commits.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-02 18:17:01 +02:00
Takashi Sakamoto 96e5fbb0dd ALSA: firewire-tascam: add proc node to show firmware information
TASCAM FireWire series has certain registers for firmware information.

This commit adds proc node to show the information.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-02 18:17:01 +02:00
Takashi Sakamoto 6f81ba19ac ALSA: firewire-tascam: add a structure for model-dependent parameters.
TASCAM FireWire series doesn't tell drivers their capabilities, thus
the drivers should have model-dependent parameters and apply it to
detected devices.

This commit adds a structure to represent such parameters.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-02 18:17:00 +02:00
Takashi Sakamoto c0949b2785 ALSA: firewire-tascam: add skeleton for TASCAM FireWire series
This commit adds a new driver for TASCAM FireWire series. In this commit,
this driver just creates/removes card instance according to bus event.
More functionalities will be added in following commits.

TASCAM FireWire series consists of:
 * PDI 1394P23 for IEEE 1394 PHY layer
 * PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
 * XILINX XC9536XL
 * XILINX Spartan-II XC2S100
 * ATMEL AT91M42800A

Ilya Zimnovich had investigated TASCAM FireWire series in 2011, and
discover some features of his FW-1804. You can see a part of his research
in FFADO project.
http://subversion.ffado.org/wiki/Tascam

A part of my work are based on Ilya's investigation, while this series
doesn't support the FW-1804, because of a lack of config ROM
information and its protocol detail, especially for PCM channels.

I observed that FW-1884 and FW-1082 don't work properly with 1394 OHCI
controller based on VT6315. The controller can actually communicate packets
to these models, while these models generate no sounds. It may be due to
the PHY/LINK layer issues. Using 1394 OHCI controller produced by the other
vendors such as Texas Instruments may work. Or adding another node on the
bus.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-02 18:16:59 +02:00
Yitian Bu 4873867e5f ASoC: dwc: correct irq clear method
from Designware I2S datasheet, tx/rx XRUN irq is cleared by
reading register TOR/ROR, rather than by writing into them.

Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
2015-10-02 11:44:04 +01:00
Rick Mann 2d1180e369 ASoC: tlv320aic3x: Prevent writing reserved registers on tlv320aic3104 CODECs
The current code writes a set of registers that are reserved on the
tlc320aic3104. The change skips those registers for that IC.

Signed-off-by: Rick Mann <rmann@latencyzero.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-01 11:15:45 +01:00
Oder Chiou 8c1a9d6323 ASoC: rt5645: Correct the naming and setting of ADC Boost Volume Control
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-01 11:13:04 +01:00
Oder Chiou e29fd55d8d ASoC: rt5645: Add the control of ClassD modulator Speaker Gain Ratio
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-01 11:12:45 +01:00
Daniel Mack 58ceb57ec1 ASoC: pxa: pxa-pcm-lib: switch over to snd-soc-dmaengine-pcm
This patch removes the old PXA DMA API usage and switches over to
generic functions provided by snd-soc-dmaengine-pcm.

More cleanups may be done on top of this, and some function stubs can
now be removed completetly. However, the intention here was to keep
the transition as small as possible.

This was tested on the mioa701 pxa27x board.

Signed-off-by: Daniel Mack <zonque@gmail.com>
[trivial change from mmp-dma to pxa-dma]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-30 23:21:16 +01:00
Mark Brown 5c0e869357 ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
 very widely spread over many different drivers.  Nothing especially
 stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' into asoc-pxa

ASoC: Fixes for v4.3

A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers.  Nothing especially
stands out, it's mostly all device specific and relatively minor.
2015-09-30 23:21:11 +01:00
Gianluca Renzi e256da84a0 ASoC: sgtl5000: fix wrong register MIC_BIAS_VOLTAGE setup on probe
Signed-off-by: Gianluca Renzi <gianlucarenzi@eurekelettronica.it>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
2015-09-30 19:44:03 +01:00
Maxime Ripard debb972487 ASoC: sun4i-codec: Remove redundant SND_PCM_RATE_KNOT
The SND_PCM_RATE_KNOT covers all the rate settings, even though some that
we don't support, while we also list all the rate we support. Simply remove
it.

Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-30 19:41:14 +01:00
Maxime Ripard 98a4b665bc ASoC: sun4i-codec: pass through clk_set_rate error
The current code, disregarding the clk_set_rate error code, was always
returning -EINVAL. Fix that and return the code in order to have more clue
about what's going on.

Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-30 19:41:14 +01:00
Jiada Wang 57622aef86 ASoC: wm8962: balance pm_runtime_enable
pm_runtime_enable is called in probe to enable runtime PM
for wm8962 codec, but pm_runtime_disable isn't called in remove
callback, nor is called in error path if probe fails after runtime
PM is enabled, this causes unbalanced pm_runtime_enable.

This patch Adds pm_runtime_disable in remove callback and error path,
to balance pm_runtime_enable.

Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-30 19:10:58 +01:00
Takashi Sakamoto 44b7308871 ALSA: firewire-digi00x: add support for asynchronous messaging
Digi 002/003 family uses asynchronous transaction for messaging.
The address to transmit this message is stored on a certain register.

This commit allocates a range of address on OHCI 1394 host controller
to handle the messaging. As long as I know, the purpose of this message
seems to notify lost of synchronization. While, the meaning of content
of the message is not clear.

Actual examples of this messaging:
 * When clock source is set as internal:
 - 0x00007051
 - 0x00007052
 - 0x00007054
 - 0x00007057
 - 0x00007058
 * When clock source is set as somewhat external:
 - 0x00009000
 - 0x00009010
 - 0x00009020
 - 0x00009021
 - 0x00009022

The lost often occurs when using internal clock source. In this case,
users hear sounds with quite short gap every several minutes. In fact,
the lost is recovered temporarily.

When using with external clock source, the lost seems not to occur. The
mechanism is not clear yet.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-30 15:37:55 +02:00
Takashi Sakamoto 660dd3d52e ALSA: firewire-digi00x: 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>
2015-09-30 15:36:56 +02:00
Takashi Sakamoto 0120d0f1fe ALSA: firewire-digi00x: add PCM functionality
This commit adds PCM functionality to transmit/receive PCM samples.

Any PCM substreams are jointed because incoming/outgoing AMDTP streams
are bound. When one of PCM substream is running or external clock source
is selected, current sampling rate is used. Else, the sampling rate is
changed as an userspace application requests.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-30 15:36:40 +02:00
Takashi Sakamoto 927f17dc3a ALSA: firewire-digi00x: add proc node to show clock status
This commit adds proc node to show current clock status for debugging.

As long as testing Digi 002 rack, registers can show local clock rate,
local clock source. When external clock input such as S/PDIF is
connected, the registers show the detection and external clock rate.

Additionally, the registers show the mode of optical digital input
interface. Although, a tester with Digi 003 rack reports this makes no
sense. Further investigation is required for Digi 003 series.

Besides, in Digi 002 rack, the S/PDIF format must be IEC 60958-4,
so-called professional.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-30 15:35:31 +02:00
Takashi Sakamoto 3a2a17974e ALSA: firewire-digi00x: add stream functionality
This commit adds a functionality to manage streaming.

The streaming is not controlled by CMP in IEC 61883-6. It's controlled by
IEEE 1394 write transaction to certain addresses.

Several clock sources are available, while there're no differences about
packet transmission. The value of SYT field in transmitted packets is
always zero. Thus, streams in both direction don't build synchronization.

And the device always requires received packets to transmit packets. This
driver keeps to transfer outgoing stream even if they're not required.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-30 15:34:55 +02:00
Takashi Sakamoto 163ae6f3f3 ALSA: firewire-digi00x: add data block processing layer
Digi 002/003 family uses its own format for data blocks. The format is
quite similar to AM824 in IEC 61883-6, while there're some differences:

 * The Valid Bit Length (VBL) code is always 0x40 in Multi-bit Linear Audio
   (MBLA) data channel.
 * The first data channel includes MIDI messages, against IEC 61883-6
   recommendation.
 * The Counter field is always zero in MIDI conformant data channel.
 * Sequence multiplexing in IEC 61883-6 is not applied to the MIDI
   conformant data channel.
 * PCM samples are scrambled in received AMDTP packets. We call the way
   as Double-Oh-Three (DOT). The algorithm was discovered by
   Robin Gareus and Damien Zammit in 2012.

This commit adds data processing layer to satisfy these differences.

There's a quirk about transmission mode for received packets. When this
driver applies non-blocking mode to outgoing packets with isochronous
channel 2 or more, after 15 to 20 seconds since playbacking, any PCM
samples causes noisy sound on the device. With isochronous channel 0 or 1,
this doesn't occur. As long as I investigated, this quirk is not observed
when applying blocking mode to the received packets.

This driver applies blocking mode to outgoing packets, while non-blocking
mode to incoming packgets.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-30 15:34:25 +02:00
Takashi Sakamoto 9edf723fd8 ALSA: firewire-digi00x: add skeleton for Digi 002/003 family
This commit adds a new driver for Digidesign 002/003 family. This commit
just creates/removes card instance according to bus event. More functions
will be added in following commits.

Digidesign 002/003 family consists of:
 * Agere FW802B for IEEE 1394 PHY layer
 * PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
 * ALTERA ACEX EP1K50 for IEC 61883 layer and DSP controller
 * ADSP-21065L for signal processing

[minor cleanup using skip_spaces() by tiwai]

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-30 15:33:50 +02:00
Dave Airlie 2d4df13c0f Merge tag 'topic/drm-misc-2015-09-25' of git://anongit.freedesktop.org/drm-intel into drm-next
Another attempt at drm-misc for 4.4 ...
- better atomic helpers for runtime pm drivers
- atomic fbdev
- dp aux i2c STATUS_UPDATE handling (for short i2c replies from the sink)
- bunch of constify patches
- inital kerneldoc for vga switcheroo
- some vblank code cleanups from Ville and Thierry
- various polish all over

* tag 'topic/drm-misc-2015-09-25' of git://anongit.freedesktop.org/drm-intel: (57 commits)
  drm/irq: Add drm_crtc_vblank_count_and_time()
  drm/irq: Rename drm_crtc -> crtc
  drm: drm_atomic_crtc_get_property should be static
  drm/gma500: Remove DP_LINK_STATUS_SIZE redefinition
  vga_switcheroo: Set active attribute to false for audio clients
  drm/core: Preserve the fb id on close.
  drm/core: Preserve the framebuffer after removing it.
  drm: Use vblank timestamps to guesstimate how many vblanks were missed
  drm: store_vblank() is never called with NULL timestamp
  drm: Clean up drm_calc_vbltimestamp_from_scanoutpos() vbl_status
  drm: Limit the number of .get_vblank_counter() retries
  drm: Pass flags to drm_update_vblank_count()
  drm/i915: Fix vblank count variable types
  drm: Kill pixeldur_ns
  drm: Stop using linedur_ns and pixeldur_ns for vblank timestamps
  drm: Move timestamping constants into drm_vblank_crtc
  drm/fbdev: Update legacy plane->fb refcounting for atomic restore
  drm: fix kernel-doc warnings in drm_crtc.h
  vga_switcheroo: Sort headers alphabetically
  drm: Spell vga_switcheroo consistently
  ...
2015-09-30 08:35:45 +10:00
Benoît Thébaudeau b7b01d345b ASoC: imx-ssi: Fix DAI hardware signal inversions
SND_SOC_DAIFMT_{IB|NB}_{IF|NF} are defined as inverting or not BCLK or
FRM relatively to what is standard for the specified DAI hardware audio
format. Consequently, the absolute polarities of these signals cannot be
derived only from these settings as this driver did. The format has to
be taken into account too.

This fixes inverted left/right channels in I²S mode.

Signed-off-by: Benoît Thébaudeau <benoit@wsystem.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-29 17:28:33 +01:00
Gianluca Renzi fb97d75b03 ASoC: sgtl5000: fix error message output for MicBias voltage
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Fabio Estevam <fabio.estevam@freescale.com>
Cc: alsa-devel@alsa-project.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Gianluca Renzi <gianlucarenzi@eurekelettronica.it>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-29 17:06:22 +01:00
kbuild test robot 02e6ef9fba ALSA: firewire-lib: process_rx_data_blocks() can be static
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 16:52:52 +02:00
Takashi Iwai 094435d41d ALSA: pcm: Avoid double hw_free calls at releasing a stream
snd_pcm_release_substream() always calls hw_free op when the stream
was opened.  This is superfluous in most cases because it's been
already released via explicit hw_free ioctl.  Although this double
call is usually OK as this callback should be written to be called
multiple times, it's better to avoid superfluous calls.

Reported-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja Kp <jeeja.kp@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:57:42 +02:00
Takashi Iwai d8dfacf8b5 Merge branch 'topic/firewire-update' into for-next 2015-09-29 12:52:07 +02:00
Takashi Sakamoto df075feefb ALSA: firewire-lib: complete AM824 data block processing layer
This commit moves the codes related to data block processing from packet
streaming layer to AM824 layer.

Each driver initializes amdtp stream structure for AM824 data block by
calling amdtp_am824_init(). Then, a memory block is allocated for AM824
specific structure. This memory block is released by calling
amdtp_stream_destroy().

When setting streaming parameters, it calls amdtp_am824_set_parameters().
When starting packet streaming, it calls amdtp_stream_start(). When
stopping packet streaming, it calls amdtp_stream_stop().

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:51:20 +02:00
Takashi Sakamoto 49c7b3fcd9 ALSA: firewire-lib: rename macros with AM824 prefix
This commit renames some macros just related to AM824 format. In later
commit, they're moved to AM824 layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:50:23 +02:00
Takashi Sakamoto 85130cb43e ALSA: firewire-lib: rename PCM format helper function
Setting the format of PCM substream to AMDTP stream structure is important
to set a handler to copy actual PCM samples between buffers. The
processing should be in data block processing layer because essentially
it has no relationship to packet streaming.

This commit renames PCM format setting function to prepare for integrating
AM824 layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:49:04 +02:00
Takashi Sakamoto 03e2a67eed ALSA: firewire-lib: move MIDI trigger helper function to AM824 layer
In IEC 61883-6, MIDI messages are transferred in MIDI conformant data
channel. Essentially, packet streaming layer is not responsible for MIDI
functionality.

This commit moves MIDI trigger helper function from the layer to AM824
layer. The rest of codes related to MIDI functionality will be moved in
later commits.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:49 +02:00
Takashi Sakamoto f65be911c0 ALSA: firewire-lib: add helper functions to set positions of data channels
In IEC 61883-6, several types of data are available in AM824 format. The
data is transferred in each data channel. The position of data channel in
data block differs depending on model.

Current implementation has an array to map the index of data channel in an
data block to the position of actual data channel. The implementation
allows each driver to access the mapping directly.

In later commit, the mapping is in specific structure pushed into an
opaque pointer. Helper functions are required.

This commit adds the helper functions for this purpose. In IEC 61883-6,
AM824 format supports many data types, while this specification easily
causes over-engineering. Current AM824 implementation is allowed to handle
two types of data, Multi Bit Linear Audio data (=PCM samples) and MIDI
conformant data (=MIDI messages).

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:36 +02:00
Takashi Sakamoto bc8500da3e ALSA: firewire-lib: move PCM substream constraint to AM824 layer
In IEC 61883-6, PCM frames are transferred in Multi Bit Linear Audio data
channel. The data channel transfers 16/20/24 bit PCM samples. Thus, PCM
substream has a constrain about it.

This commit moves codes related to the constraint from packet streaming
layer to AM824 data block processing layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:19 +02:00
Takashi Sakamoto 51c29fd213 ALSA: firewire-lib: rename parameter setting function for AM824 with FDF field
The value of FDF field in CIP header is protocol-dependent. Thus, it's
better to allow data block processing layer to decide the value in any
timing.

In AM824 data format, the value of FDF field in CIP header indicates
N-flag and Nominal Sampling Frequency Code (sfc). The N-flag is for
switching 'Clock-based rate control mode' and 'Command-based rate control
mode'. In our implementation, 'Clock-based rate control mode' is just
supported. Therefore, When sampling transfer frequency is decided, then
the FDF can be set.

This commit replaces 'amdtp_stream_set_parameters' with
'amdtp_am824_set_parameters' to set the FDF. This is the same timing
to decide the ration between the number of data blocks and the number of
PCM frames.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:48:02 +02:00
Takashi Sakamoto 5955815e71 ALSA: firewire-lib: add data block processing layer for AM824 format
This commit adds data block processing layer for AM824 format. The new
layer initializes streaming layer with its value for fmt field.

Currently, most implementation of data block processing still remains
streaming layer. In later commits, these codes will be moved to the layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:47:45 +02:00
Takashi Sakamoto d67c46b953 ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' to prepare for functional separation
In later commit, data block processing layer will be newly added. This
layer will be named as 'amdtp-am824'.

This commit renames current amdtp file to amdtp-stream, to distinguish it
from the new layer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:47:11 +02:00
Takashi Sakamoto 414ba022a5 ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header
Some vendor specific protocol uses its own value for fmt/fdf fields in
CIP header.

This commit support to set arbitrary values for the fields.

In IEC 61883-6, NO-DATA code is defined for FDF field. A packet with this
code includes no data even if it includes some data blocks. This commit
still leaves a condition to handle this special packet.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:48 +02:00
Takashi Sakamoto 20e445771e ALSA: firewire-lib: add helper functions as interfaces between packet streaming layer and data block processing layer
ALSA PCM framework uses PCM buffer with a concept of 'period' to
synchronize userspace operations to hardware for nearly-realtime
processing. Each driver implements snd_pcm_period_elapsed() to tell across
of the period boundary to ALSA PCM middleware. To call the function, some
drivers utilize hardware interrupt handlers, the others count handled PCM
frames.

Drivers for sound units on IEEE 1394 bus are the latter. They use two
buffers; PCM buffer and DMA buffer for IEEE 1394 isochronous packet. PCM
frames are copied between these two buffers and 'amdtp_stream' structure
counts the handled PCM frames. Then, snd_pcm_period_elapsed() is called if
required.

Essentially, packet streaming layer should not be responsible for PCM
frame processing. The PCM frame processing should be handled in each data
block processing layer as a result of handling data blocks. Although, PCM
frame counting is a common work for all of protocols which ALSA firewire
stack is going to support.

This commit adds two new helper functions as interfaces between packet
streaming layer to data block processing layer. In future, each data block
processing layer implements these functions. The packet streaming layer
calls data block processing layer per packet by calling the functions. The
data block processing layer processes data blocks and PCM frames, and
returns the number of processed PCM frames. Then the packet streaming layer
calculates handled PCM frames and calls snd_pcm_period_elapsed().

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:47 +02:00
Takashi Sakamoto 6a4e89ff0f ALSA: firewire-lib: add a member of frame_multiplier instead of double_pcm_frames
In future commit, interface between data block processing layer and packet
stream processing layer is defined. These two layers communicate the
number of data blocks and the number of PCM frames.

The data block processing layer has a responsibility for calculating the
number of PCM frames. Therefore, 'dual wire' of Dice quirk should be
handled in data block processing layer.

This commit adds a member of 'frame_multiplier'. This member represents
the ratio of the number of PCM frames against the number of data blocks.
Usually, the value of this member is 1, while it's 2 in Dice's 'dual wire'.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:46 +02:00
Takashi Sakamoto 27ec83b5c6 ALSA: firewire-lib: add an argument for Dice's dual wire mode
In IEC 61883-6, one data block represents one event. In ALSA, the event is
one PCM frame. Therefore, when processing one data block, current
implementation counts one PCM frame.

On the other hand, Dice platform has a quirk called as 'dual wire' at
higher sampling rate. In detail, see comment of commit 6eb6c81eee
("ALSA: dice: Split stream functionality into a file").

Currently, to handle this quirk, AMDTP stream structure has a
'double_pcm_frames' member. When this is enabled, two PCM frames are
counted. Each driver set this flag by accessing the structure member
directly.

In future commit, some members related to AM824 data block will be moved
to specific structure, to separate packet streaming layer and data block
processing layer. The access will be limited by opaque pointer.

For this reason, this commit adds an argument into
amdtp_stream_set_parameter() to set the flag.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:46 +02:00
Takashi Sakamoto 547e631ce3 ALSA: firewire-lib: return error code when amdtp_stream_set_parameters() detects error
Currently, amdtp_stream_set_parameters() returns no error even if wrong
arguments are given. This is not good for streaming layer because drivers
can continue processing ignoring capability of streaming layer.

This commit changes this function to return error code.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:45 +02:00
Takashi Sakamoto 10b2b6dc1a ALSA: firewire-lib: arrange structure for AMDTP stream
In later commit, some members related to AM824 data format will be moved
from AMDTP stream structure to data block structure. This commit is a
preparation for it. Additionally, current layout of AMDTP stream structure
is a bit mess by several extensions. This commit also arranges the layout.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-29 12:35:44 +02:00
Dan Carpenter e87359efca ALSA: usb-audio: harmless underflow in snd_audigy2nx_led_put()
We want to verify that "value" is either zero or one, so we test if it
is greater than one.  Unfortunately, this is a signed int so it could
also be negative.  I think this is harmless but it introduces a static
checker warning.  Let's make "value" unsigned.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-28 14:33:03 +02:00
Lars-Peter Clausen e74679b38c ASoC: db1200: Fix DAI link format for db1300 and db1550
Commit b4508d0f95 ("ASoC: db1200: Use static DAI format setup") switched
the db1200 driver over to using static DAI format setup instead of a
callback function. But the commit only added the dai_fmt field to one of
the three DAI links in the driver. This breaks audio on db1300 and db1550.

Add the two missing dai_fmt settings to fix the issue.

Fixes: b4508d0f95 ("ASoC: db1200: Use static DAI format setup")
Reported-by: Manuel Lauss <manuel.lauss@gmail.com>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
2015-09-25 09:31:09 -07:00
Takashi Iwai 010cf269a4 Merge branch 'topic/drm-sync-audio-rate' into for-next 2015-09-25 10:07:55 +02:00
Libin Yang ddd621fbba ALSA: hda - display audio call sync_audio_rate callback
For display audio, call the sync_audio_rate callback function
to do the synchronization between gfx driver and audio driver.

Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-25 10:04:59 +02:00
Takashi Iwai 0b2c8c12f9 ASoC: Updates for v4.4
A first batch of updates targetted at v4.4.  There are no substantial
 core fixes here, the biggest block of changes is updates to the rcar
 drivers and the addition of a CODEC driver for the AK4613.
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Merge tag 'asoc-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for v4.4

A first batch of updates targetted at v4.4.  There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
2015-09-25 08:33:11 +02:00
Takashi Iwai 7f57d803ee ALSA: hda - Disable power_save_node for Thinkpads
Lenovo Thinkpads with recent Realtek codecs seem suffering from click
noises at power transition since the introduction of widget power
saving in 4.1 kernel.  Although this might be solved by some delays in
appropriate points, as a quick workaround, just disable the
power_save_node feature for now.  The gain it gives is relatively
small, and this makes the situation back to pre 4.1 time.

This patch ended up with a bit more code changes than usual because
the existing fixup for Thinkpads is highly chained.  Instead of adding
yet another chain, combine a few of them into a single fixup entry, as
a gratis cleanup.

Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 20:49:00 +02:00
Takashi Iwai 1ce3cbe2ab ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
 very widely spread over many different drivers.  Nothing especially
 stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v4.3

A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers.  Nothing especially
stands out, it's mostly all device specific and relatively minor.
2015-09-24 20:48:01 +02:00
Lukas Wunner 21b45676b7 vga_switcheroo: Set active attribute to false for audio clients
The active attribute in struct vga_switcheroo_client denotes whether
the outputs are currently switched to this client. The attribute is
only meaningful for vga clients. It is never used for audio clients.

The function vga_switcheroo_register_audio_client() misuses this
attribute to store whether the audio device is fully initialized.
Most likely there was a misunderstanding about the meaning of
"active" when this was added.

Comment from Takashi's review:

"Not really.  The full initialization of audio was meant that the audio
is active indeed.  Admittedly, though, the active flag for each audio
client doesn't play any role because the audio always follows the gfx
state changes, and the value passed there doesn't reflect the actual
state due to the later change.  So, I agree with the removal of the
flag itself -- or let the audio active flag following the
corresponding gfx flag.  The latter will make the proc output more
consistent while the former is certainly more reduction of code."

Set the active attribute to false for audio clients. Remove the
active parameter from vga_switcheroo_register_audio_client() and
its sole caller, hda_intel.c:register_vga_switcheroo().

vga_switcheroo_register_audio_client() was introduced by 3e9e63dbd3
("vga_switcheroo: Add the support for audio clients"). Its use in
hda_intel.c was introduced by a82d51ed24 ("ALSA: hda - Support
VGA-switcheroo").

v1.1: The changes above imply that in find_active_client() the call
to client_is_vga() is now superfluous. Drop it.

Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Lukas Wunner <lukas@wunner.de>
[danvet: Add Takashi's clarification to the commit message.]
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
2015-09-24 20:14:25 +02:00
Axel Lin 0f7e177406 ASoC: rl6347a: Clean up unneeded inclusion of header files
Also move the include of sound/hda_verbs.h to rl6347a.h because it is used
in rl6347a.h.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-24 10:30:47 -07:00
Takashi Iwai db25f440f4 Merge branch 'for-linus' into for-next 2015-09-24 14:02:48 +02:00
Takashi Iwai 83510441bc ALSA: hda/tegra - async probe for avoiding module loading deadlock
The Tegra HD-audio controller driver causes deadlocks when loaded as a
module since the driver invokes request_module() at binding with the
codec driver.  This patch works around it by deferring the probe in a
work like Intel HD-audio controller driver does.  Although hovering
the codec probe stuff into udev would be a better solution, it may
cause other regressions, so let's try this band-aid fix until the more
proper solution gets landed.

Reported-by: Thierry Reding <treding@nvidia.com>
Tested-by: Thierry Reding <treding@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 14:01:46 +02:00
Dan Carpenter 145d92e77e ALSA: core: check for underflow in snd_pcm_sw_params()
As far as I can see, having an invalid ->tstamp_mode is harmless, but
adding a check silences a static checker warning.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 11:46:25 +02:00
Takashi Sakamoto 13f3a46d2a ALSA: oxfw: add Mackie Onyx Satellite quirk to inform wrong value in 'dbs' field in tx CIP header
Mackie Onyx Satellite has two usage; standalone and with base station.

These two modes has different stream formats. In standalone mode, rx data
block includes 2 Multi Bit Linear Audio (MBLA) data channels and tx data
block includes 2. With base station, they're 6 and 2.

Although, with base station, the actual tx packet include wrong value in
'dbs' field in its CIP header. This quirk causes packet streaming layer to
detect packet discontinuity and to stop PCM substream.

This is a response of 'single' subfunction 'extended stream format
information' command in AV/C Stream Format Information Specification 1.1.
It means that a data block in tx packets includes 2 MBLA data channels.

$ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffffffff
response: 000: 0c ff bf c0 01 00 00 00 ff 00 90 40 03 02 01 02
response: 010: 06

Current OXFW driver parses the response and detects stream formats
correctly.

$ cat /proc/asound/card1/firewire/formation
...
Output Stream from device:
	Rate	PCM	MIDI
* 	48000	2	0
 	44100	2	0
 	88200	2	0
	96000	2	0

On the other hand, in actual tx CIP, the 'dbs' field has 6. But the number
of quadlets in CIP payload is not multiple of 6. Seeing the subtraction of
two successive payload quadlets, it should be multiple of 2.

payload  CIP      CIP
quadlets header0  header1
      24 00060052 9002ffff
      24 0006005e 9002ffff
      26 0006006a 9002ffff
      24 00060077 9002ffff
      24 00060083 9002ffff
      26 0006008f 9002ffff
      24 0006009c 9002ffff
      24 000600a8 9002ffff
      26 000600b4 9002ffff
      24 000600c1 9002ffff

This commit adds support for this quirk to OXFW driver, by using
CIP_WRONG_DBS flag. When this flag is set, packet streaming layer uses
the value of its 'data_block_quadlets' member instead of the value in CIP
header. This value is already set by OXFW driver and no discontinuity is
detected.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-24 11:44:25 +02:00
Mark Brown 4922e7a1b7 Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next 2015-09-23 11:01:30 -07:00
Mark Brown 727e53a12f Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/spear', 'asoc/topic/sunxi' and 'asoc/topic/tlv320aic3x' into asoc-next 2015-09-23 11:01:27 -07:00
Mark Brown e32efed3ea Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar' and 'asoc/topic/rockchip' into asoc-next 2015-09-23 11:01:25 -07:00
Mark Brown 9abbe73e7d Merge remote-tracking branches 'asoc/topic/jz4740', 'asoc/topic/kirkwood', 'asoc/topic/mtk', 'asoc/topic/multi' and 'asoc/topic/mxs' into asoc-next 2015-09-23 11:01:22 -07:00
Mark Brown f7b93159ab Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next 2015-09-23 11:01:19 -07:00
Mark Brown 97e15d9a8b Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next 2015-09-23 11:01:16 -07:00
Mark Brown f950aa96d1 Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next 2015-09-23 11:01:15 -07:00
Mark Brown ed14ee0eea Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus 2015-09-23 11:01:12 -07:00
Mark Brown d86a2fe4dc Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/psc', 'asoc/fix/pxa', 'asoc/fix/spear', 'asoc/fix/sti' and 'asoc/fix/wm0010' into asoc-linus 2015-09-23 11:01:08 -07:00
Mark Brown 312e0bce7f Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/doc', 'asoc/fix/fsl-card', 'asoc/fix/fsl-ssi', 'asoc/fix/intel' and 'asoc/fix/maintainers' into asoc-linus 2015-09-23 11:01:03 -07:00
Mark Brown fcffa0dbb6 Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus 2015-09-23 11:01:02 -07:00
Mark Brown 5ee84ba8c7 Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus 2015-09-23 11:01:00 -07:00
Oder Chiou fce97b4d70 ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:59:13 -07:00
Oder Chiou 4f4794124e ASoC: rt5645: Increase the delay time to remove the pop sound
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:59:10 -07:00
Oder Chiou 21cb13e72b ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:59:07 -07:00
Oder Chiou de3f8fdf73 ASoC: rt5645: Remove the repeated definitions
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-23 10:57:52 -07:00
Robert Jarzmik 8811191fdf ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
PCM receive and transmit DMA requestor lines were reverted, breaking the
PCM playback interface for PXA platforms using the sound/soc/ variant
instead of the sound/arm variant.

The commit below shows the inversion in the requestor lines.

Fixes: d65a14587a ("ASoC: pxa: use snd_dmaengine_dai_dma_data")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
2015-09-22 14:12:46 -07:00
John Lin 917536aeb8 ASoC: rt5645: Add jd_invert for Broadwell
Broadwell can not triger the IRQ falling and rising simultaneously,
so it can not detect jack-in and jack-out simultaneously.
We add a flag "jd_invert" to platform data. If this flag is set,
codec IRQ will be set to invert that forces IRQ as pulse when jack-in
and jack-out.

Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-22 09:46:33 -07:00
John Lin e9159e7577 ASoC: rt5645: Add dmi for Broadwell
Add DMI data for Buddy project.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-22 09:46:32 -07:00
Mark Brown c44506997c Merge branch 'fix/rt5645' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-rt5645 2015-09-22 09:45:46 -07:00
Adam Thomson 5b64c173cd ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
Current code incorrectly treats dai format for AC97 as bit mask
whereas it's actually an integer value. This causes DAI formats
other than AC97 (e.g. DSP_B) to trigger AC97 related code,
which is incorrect and breaks functionality. This patch fixes
the code to correctly compare values to determine AC97 or not.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 11:07:42 -07:00
Ricard Wanderlof cde79035c6 ASoC: Handle multiple codecs with split playback / capture
Add the capability to use multiple codecs on the same DAI linke where
one codec is used for playback and another one is used for capture.

Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS
microphone for capture.

No verification is made that the playback and capture codec setups are
compatible in terms of number of TDM slots (where applicable).

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 10:58:04 -07:00
Jyri Sarha 9529138276 ASoC: hdmi: Remove obsolete dummy HDMI codec
The hdmi stub codec has not been used since refactoring of OMAP HDMI
audio support.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 10:44:39 -07:00
Jeeja KP 5b97c0f18a ASoC: Intel: Skylake: Remove unused CPU dais
We need to create CPU DAI for each endpoint instance. For this we
should have one DMIC DAI, one HDA DAI and SSP DAI.  Thus, DMIC23,
HDA-SPK/AMIC was not required so this patch removes them

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 09:27:52 -07:00
Sudip Mukherjee f072f91aa7 ASoC: wm0010: fix error path
Fix the error path so that we can free the allocated memory on the error
path instead of releasing them individually on each error.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 08:40:55 -07:00
Sudip Mukherjee 2ace47be5a ASoC: wm0010: fix memory leak
We have requested for the firmware but we have missed releasing it both
on success and on error path.
While checking the code it turned out that the requested firmware is not
even used. More over the same firmware is being loaded by
wm0010_stage2_load().

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 08:40:50 -07:00
Zidan Wang 8524bb0c7a ASoC: wm8960: correct the max register value of mic boost pga
the max register value of mic boost pga should be 3.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 07:51:09 -07:00
Zidan Wang ee92cfb030 ASoC: wm8962: remove 64k sample rate support
wm8962 can't support 64k sample rate. When playing a 64KHz wave file,
'Unsupported rate 64000Hz' will be prompted.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 07:50:23 -07:00
Axel Lin 9d70594760 ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency
Make SND_SOC_ROCKCHIP_MAX98090 and SND_SOC_ROCKCHIP_RT5645 depend on
CLKDEV_LOOKUP to fix below build warning:

warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects
SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K &&
!UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP)

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 07:47:57 -07:00
Kuninori Morimoto e3a4d958d7 ASoC: ak4613: add Digital Playback Volume support
For L/ROUT1 to L/ROUT6

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-19 07:41:15 -07:00
Peter Ujfalusi ab1fffe3a7 ASoC: davinci-mcasp: Fix devm_kasprintf format string
The '\n' at the end of the format string is not needed. It adds an extra
line break when doing
cat /proc/interrupts

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-18 16:22:07 +01:00
Kuninori Morimoto b075706284 ASoC: add ak4613 support
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-17 12:33:08 +01:00
kbuild test robot 31f4f0c640 ASoC: sunxi: fix platform_no_drv_owner.cocci warnings
sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it.

 Remove .owner field if calls are used which set it automatically

Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci

CC: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-17 11:38:03 +01:00
Jyri Sarha 14a998be08 ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data
The slot_width is for essentially same thing. Instead of storing
bclk_lrclk_ratio, just store the slot_width. Comments has been updated
accordingly and some variable names changed to more descriptive.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-17 11:37:27 +01:00
Julia Lawall 8985729578 ASoC: qcom: change PTR_ERR argument
Apply PTR_ERR to the value that was recently assigned.

The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)

// <smpl>
@@
expression x,y;
@@

if (IS_ERR(x) || ...) {
  ... when any
      when != IS_ERR(...)
(
   PTR_ERR(x)
|
*  PTR_ERR(y)
)
  ... when any
}
// </smpl>

Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-17 11:36:19 +01:00
Jyri Sarha 6c84e5917f ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
Store return value of of_get_property() to a pointer of __be32 type as
device tree has big endian type. This fixes a sparse warning couple of
lines later when be32_to_cpup() is used to convert from big endian to
cpu endian.

The whole conversion is not really necessary, as we are only checking
if the value is zero or not, but I wanted to add it to remind in the
future that the data has to be converted before use. Compiler should
optimize the unnecessary operations away.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-17 11:36:05 +01:00
Charles Keepax ccaadda319 ASoC: wm5110: Add additional analogue input enable for early revs
Earlier revisions of the wm5110/8280 silicon require a slightly more
complex procedure to enable analogue inputs. This patch adds this into
the driver.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-17 11:35:02 +01:00
Charles Keepax 002b083b8d ASoC: arizona: Add utility function to check if an input is analog
We will occasionally require to take different action based on if an
input is analog or digital so add a helper function to return if an
input is analog.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 21:00:03 +01:00
Charles Keepax bee261b896 ASoC: arizona: Add default cases for event switches
Since the addition of the WILL_PMU / WILL_PMD several of the switches in
arizona.c no longer cover all cases or have a default case. Whilst this
isn't causing any problems in the interests of robustness add default
cases.

Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 21:00:03 +01:00
Robert Jarzmik 3c8f7710c1 ASoC: fix broken pxa SoC support
The previous fix of pxa library support, which was introduced to fix the
library dependency, broke the previous SoC behavior, where a machine
code binding pxa2xx-ac97 with a coded relied on :
 - sound/soc/pxa/pxa2xx-ac97.c
 - sound/soc/codecs/XXX.c

For example, the mioa701_wm9713.c machine code is currently broken. The
"select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for
compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared
in sound/arm/Kconfig and sound/soc/pxa/Kconfig.

Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct
pxa2xx-ac97 compilation.

Fixes: 846172dfe3 ("ASoC: fix SND_PXA2XX_LIB Kconfig warning")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 20:58:27 +01:00
Kuninori Morimoto 462c30bc8b ASoC: rsnd: fixup clkout_name[] index error for single AUDIO_CLKOUT SoC
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") added AUDIO_CLKOUT
support for ADG. But single AUDIO_CLKOUT needs clkout_name[CLKOUT],
not clkout_name[i]. Kernel will have NULL pointer access without this
patch.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 20:40:39 +01:00
Kuninori Morimoto 544637bf3f ASoC: ak4642: exchange macro name to avoid redefinition
71a0138ab("ASoC: ak4642: enable to use MCKO as fixed rate output
pin on DT") added new FS() macro, but x86 already has it in
arch/x86/include/uapi/asm/ptrace-abi.h
This patch exchange FS() to FSs() to avoid redefinition warning

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 20:40:27 +01:00
Kuninori Morimoto e2f8445176 ASoC: rsnd: SND_SOC_RCAR depends on COMMON_CLK
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") uses
of_clk_add_provider() which is requesting struct clk_onecell_data.
But it is COMMON_CLK feature. SND_SOC_RCAR depends on COMMON_CLK

This patch also solved compile error of 7486d80f7("ASoC: rsnd: remove
unneeded sh_clk header")

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 20:40:02 +01:00
Emilio López 45fb6b6f2a ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs
The sun4i, sun5i and sun7i SoC families have a built-in codec, capable
of both audio capture and playback.

While this is called a codec by Allwinner, it really is an in-SoC
combination of a codec and a DAI, with its own DAC/ADC and amplifiers
in a single memory-mapped controller.

The capture part has been left out for now, and will be added eventually.

Signed-off-by: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 20:18:45 +01:00
Sudip Mukherjee 2fc171e69e ASoC: Intel: remove unused function
The function get_current_pipe_id() was not being used.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 18:30:47 +01:00
Jyri Sarha 6131084a0b ASoC: simple-card: Add tdm slot mask support to simple-card
Adds DT binding for explicitly choosing a tdm mask for DAI and uses it
in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been
changed.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 18:00:30 +01:00
Jyri Sarha dd55ff8346 ASoC: davinci-mcasp: Add set_tdm_slots() support
Implements set_tdm_slot() callback for mcasp. Channel constraints are
updated according to the configured tdm mask and slots each time
set_tdm_slot() is called. The special case when slot width is set to
zero is allowed and it means that slot width is the same as the sample
width.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 17:59:15 +01:00
Jyri Sarha 3e8f5263bd ASoC: tlv320aic3x: Improve tdm support
Before this patch the set_tdm_slots() callback did not store the value
of slot width anywhere. The tdm support only worked if selected slot
width was equal to the sample width. With this patch all sample widths
that fit into the slot width are supported. There unused bits are
filled unnecessarily in the capture direction, but the other end of
the i2s bus should be able to ignore them.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 17:58:56 +01:00
Axel Lin 03d3964c51 ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 17:46:11 +01:00
Mark Brown df5c79e982 Merge branch 'fix/rt5645' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fix-rt5645 2015-09-14 19:53:39 +01:00
Kuninori Morimoto 2a46db4a37 ASoC: rsnd: add AUDIO_CLKOUT support
Renesas sound has AUDIO_CLKOUT (in Gen1/Gen2) AUDIO_CLKOUT1/2/3 (in Gen3)
This patch support these patches as clock provider.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:32 +01:00
Kuninori Morimoto 248e88c2fb ASoC: rsnd: adg: ignore undefined clock error
undefined clock is not error. Accept such case. And this is prepare
for clock out support in the same time.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:32 +01:00
Kuninori Morimoto 3af6c3ac91 ASoC: rsnd: tidyup ADG debug message for clock selection
It didn't have "\n", and indicated different data

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:32 +01:00
Kuninori Morimoto eae6fff4f1 ASoC: rsnd: tidyup ADG clock calculate method
Current ADG clock calculation needs ADG and SSI settings.
Thus, SSI side clock request function depends on ADG settings.
After reconsideration,  we can close this method inside ADG.
This function uses new method. And it becomes preparation for
AUDIO_CLKOUT support.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:32 +01:00
Kuninori Morimoto 5c6901d98b ASoC: rsnd: ADG initial setup on rsnd_adg_ssi_clk_init()
ADG is special IP since it doesn't have MSTP. And now, ADG has common
mod base register access. We can now setup ADG initial setting when
probe timing. It is needed if sound card is based on AUIDO_CLK which
is used as Master clock.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:31 +01:00
Kuninori Morimoto f1df122907 ASoC: rsnd: add common mod confirm method
Renesas sound has SSI/SRC/DVC/MIX/ADG modules, and these have original
register mapping. Thus this driver is using regmap field, and each module
is using it based on each module ID.

Sometimes, each module needs other module to controlling. but current each
function is using just "mod" as parameter name. This is confusable.
For example, if SSI0 and SRC2 are connected, and if SRC module function
has bug of module access, and if it needs to control connected SSI,
SRC function will access to SSI2 (It should access to SSI0, but it uses
SRC's ID 2). This is easy to happen in current driver style.

To avoid this kind of confusable trouble, this patch adds module confirm
macro for debug purpose.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:31 +01:00
Kuninori Morimoto 1665a9e5e2 ASoC: rsnd: ADG uses mod base common method
Renesas sound has ADG IP, but it is special device.
(It is clock generater, and it doesn't need MSTP)
Thus, ADG didn't use mod base common method on rsnd driver.
But it can be confusable point. Let's use common method

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:31 +01:00
Kuninori Morimoto b76e218ae5 ASoC: rsnd: add rsnd_mod_get() macro and use it
Renesas sound driver has SSI/SRC/DVC/CTU/MIX, and these are controlled
as modules. And these module are member of each modules's private data.
It used own method to get module pointer, but Let's use common method

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:31 +01:00
Kuninori Morimoto ac37a45b0b ASoC: rsnd: Add Gen3 initial support
Renesas sound Gen3 is updated version of Gen2. We need to update
driver for it, but basically it should works as Gen2 compatible.
This is initial support for Gen3. Gen3 specific feature will be
incrementally added in the future

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:31 +01:00
Kuninori Morimoto 7486d80f7d ASoC: rsnd: remove unneeded sh_clk header
sh_clk header is not needed, and it will create confusion.
Let's remove it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:29 +01:00
Kuninori Morimoto b9bfe9d5f7 ASoC: rsnd: rename rsnd_src_pcm_new() to rsnd_src_pcm_new_gen2()
rsnd_src_pcm_new() is used only from Gen2. make it clear in function name,
and remove unneeded Gen1 check.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:28 +01:00
Kuninori Morimoto c25f20e1de ASoC: rsnd: tidyup rsnd_dma_to_xxx() position
rsnd_dma_to_xxx() macro should exist in same place

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:47:28 +01:00
Kuninori Morimoto 171a0138ab ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT
ak4642 chip can output clock via MCKO pin as audio reference clock.
This patch supports it on DT

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:41:38 +01:00
Kuninori Morimoto 2719a752b6 ASoC: ak4642: use *dev on ak4642_i2c_probe()
Let's replace &i2c->dev to dev

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:41:38 +01:00
Jyri Sarha 295c3405a8 ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver
The TX and RX direction share the same bit clock and frame sync, so
the samplerate must be the same to both directions.

Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 19:40:08 +01:00
Luis de Bethencourt c81740e005 ASoC: ux500: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:47:20 +01:00
Luis de Bethencourt cfa7a38c75 ASoC: mop500: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:47:20 +01:00
Luis de Bethencourt baafd373e9 ASoC: pxa: Fix module autoload for OF platform drivers
These platform drivers have a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:46:58 +01:00
Luis de Bethencourt 30953a8047 ASoC: kirkwood: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:46:28 +01:00
Luis de Bethencourt 27fcf913f6 ASoC: jz4740: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:46:14 +01:00
Luis de Bethencourt c759241fe2 ASoC: fsl_sai: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:44:51 +01:00
Luis de Bethencourt 5226f2340c ASoC: fsl-asoc-card: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:44:51 +01:00
Luis de Bethencourt aab7826f59 ASoC: atmel_wm8904: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:42:10 +01:00
Axel Lin 9ed747641b ASoC: intel: broadwell: Convert to devm_snd_soc_register_card
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:39:23 +01:00
Axel Lin 1ff68f5590 ASoC: sh: siu_dai: Convert to use resource managed APIs
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:39:05 +01:00
Axel Lin 921e54680a ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning
Fix below build warning:
sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe':
sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable]

Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:38:40 +01:00
Axel Lin 42617869bf ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:37:25 +01:00
Axel Lin 2fd7076a43 ASoC: pxa: Convert to devm_snd_soc_register_card
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:33:41 +01:00
Axel Lin 5b73de19fd ASoC: mxs-sgtl5000: Convert to devm_snd_soc_register_card
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:33:17 +01:00
Axel Lin 76b0d1f5e7 ASoC: blackfin: Convert to devm_snd_soc_register_card
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:32:58 +01:00
Axel Lin 2342cafe22 ASoC: au1x: Convert to devm_snd_soc_register_card
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:28:24 +01:00
Koro Chen e4fba9b5be ASoC: mediatek: Increase periods_min in capture
In capture, there is chance that hw_ptr reported at IRQ is
a little smaller than period_size due to internal AFE buffer.
In the case of ping-pong buffer:

|xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------|
                            hw_ptr < period_size

This available buffer will not be read since its size is smaller than
avail_min (which is period_size by default), and read thread continues
to sleep. If the next hw_ptr is just a little larger than buffer_size,
overrun occurs. One more period can hold the possible unread buffer.

Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 18:13:54 +01:00
Peter Ujfalusi 723831927e ASoC: davinci-mcasp: Revise the FIFO threshold calculation
The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value
for the refining should meet this requirement as well.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-14 17:29:41 +01:00
Takashi Sakamoto 0052b7dcf9 ALSA: pcm: remove structure member of 'struct snd_pcm_hwptr_log *' type because this structure had been removed
This structure was added by 4d96eb255c ('ALSA: pcm_lib - add possibility
to log last 10 DMA ring buffer positions') to store PCM pointers
information of latest 10 pointer movements (=XRUN_LOG_CNT). When
CONFIG_SND_PCM_XRUN_DEBUG is configured, 'struct snd_pcm_runtime' has
'hwptr_log' member with a pointer to the structure. When calling
xrun_log() in pcm_lib.c, the structure was allocated to the pointer.
When calling snd_pcm_detach_substream() in pcm.c, the allocated pointer
is released.

In f5914908a5 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints'),
the pointer logging is replaced with using Linux Kernel Tracepoints. The
structure was also removed, while it's just declared. The member and kfree
still remains.

This commit removes the member and related codes. I think this was
overlooked because it brings no errors/warnings to C compilers.

Fixes: f5914908a5 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-13 12:03:15 +02:00
Linus Torvalds 64d1def7d3 sound fixes for 4.3-rc1
A collection of small fixes since the last update: the HD-audio
 quirks as usual with a USB-audio fix and a trivial fix for the
 old sparc driver.
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Merge tag 'sound-fix-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes since the last update: the HD-audio quirks
  as usual with a USB-audio fix and a trivial fix for the old sparc
  driver"

* tag 'sound-fix-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Change internal PCM order
  ALSA: hda - Fix white noise on Dell M3800
  ALSA: hda - Use ALC880_FIXUP_FUJITSU for FSC Amilo M1437
  ALSA: hda - Enable headphone jack detect on old Fujitsu laptops
  ALSA: sparc: amd7930: Fix module autoload for OF platform driver
  ALSA: hda - Add some FIXUP quirks for white noise on Dell laptop.
2015-09-11 09:42:32 -07:00
Sudip Mukherjee 29d28c5060 ALSA: hdsp: fix memory leak
If the size of the firmware is less than expected size then we are
exiting with the error code but we missed releasing the firmware.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-11 17:35:11 +02:00
Zidan Wang 7e90f9b2b5 ASoC: wm8960: correct gain value for input PGA and add microphone PGA
The input PGAs have a gain range from -17.25dB to +30dB in 0.75dB steps.

The boost stage can provide additional gain. For line inputs, -12dB to
+6dB gain is available on the boost mixer. For micphone inputs, it can
provide up to +29dB additional gain from the microphone PGA.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-11 13:39:15 +01:00
Zidan Wang 3758ff5f3d ASoC: wm8960: correct the min gain value of some PGA
The min gain is the corresponding gain value when the register value is 0
instead of 1, just correct it.

Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-11 13:38:53 +01:00
Oder Chiou f17b329b73 ASoC: rt5645: Remove incorrect settings
The patch removes the incorrect settings to avoid the pop sound in the
first playback with headphone after boot.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-11 12:26:51 +01:00
Sudip Mukherjee 75881df3fd ASoC: dapm: fix memory leak
Incase of an unknown event we were directly returning but we missed
freeing params.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-11 12:15:52 +01:00
Arnaud Pouliquen 3a0e27d84b ASoC: sti: check return of of_property_read
Add check on of_property_read to return error when
DT required property is not defined.

Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-10 11:11:18 +01:00
Liam Girdwood f1ec5ec7a9 ASoC: intel: Fix SSP port configuration after RTD3 resume.
Currently the SSP port settings are being clobbered as part of the DSP
RTD3 restore logic. make sure we save the correct params and restore them
at resume. The FW sadly does not save SSP settings as part of the PM
context.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-09 12:29:15 +01:00
John Lin 721b51fcf9 ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-09 12:28:54 +01:00
Johan Rastén 5ee20bc792 ALSA: usb-audio: Change internal PCM order
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.

This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.

Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf

Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-07 10:57:27 +02:00
Niranjan Sivakumar 467e1436ba ALSA: hda - Fix white noise on Dell M3800
The M3800 is very minor workstation variant of the XPS 15 which has
already been patched for this issue.  I figured it's probably more
important for this version of the laptop to be patched than the
regular XPS as Dell sells is pre-configured with Ubuntu to be used as
a Linux workstation.  I have tested the patch on my the hardware on
Linux 4.2.0.

Signed-off-by: Niranjan Sivakumar <ns253@cornell.edu>
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-05 18:22:10 +02:00
Takashi Iwai a161574e20 ALSA: hda - Use ALC880_FIXUP_FUJITSU for FSC Amilo M1437
It turned out that the machine has a bass speaker, so take a correct
fixup entry.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-05 18:18:24 +02:00
Takashi Iwai bb148bdeb0 ALSA: hda - Enable headphone jack detect on old Fujitsu laptops
According to the bug report, FSC Amilo laptops with ALC880 can detect
the headphone jack but currently the driver disables it.  It's partly
intentionally, as non-working jack detect was reported in the past.
Let's enable now.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-05 18:18:08 +02:00
Linus Torvalds abebcdfb64 sound updates for 4.3-rc1
There are little changes in core part, but lots of development are
 found in drivers, especially ASoC.  The diffstat shows regmap-
 related changes for a slight API additions / changes, and that's all.
 
 Looking at the code size statistics, the most significant addition
 is for Intel Skylake.  (Note that SKL support is still underway, the
 codec driver is missing.)  Also STI controller driver is a major
 addition as well as a few new codec drivers.
 
 In HD-audio side, there are fewer changes than the past.  The
 noticeable change is the support of ELD notification from i915
 graphics driver.  Thus this pull request carries a few changes in
 drm/i915.
 
 Other than that, USB-audio got a rewrite of runtime PM code.  It
 was initiated by lockdep warning, but resulted in a good cleanup in
 the end.
 
 Below are the highlights:
 
 Common:
 - Factoring out of AC'97 reset code from ASoC into the core helper
 - A few regmap API extensions (in case it's not pulled yet)
 
 ASoC:
 - New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
   RT298 and ST STI controllers
 - Machine drivers for Rockchip systems with MAX98090 and RT5645 and
   RT5650
 - Initial driver support for Intel Skylake devices
 - Lots of rsnd cleanup and enhancements
 - A few DAPM fixes and cleanups
 - A large number of cleanups in various drivers (conversion and
   standardized to regmap, component) mostly by Lars-Peter and Axel
 
 HD-audio:
 - Extended HD-audio core for Intel Skylake controller support
 - Quirks for Dell headsets, Alienware 15
 - Clean up of pin-based quirk tables for Realtek codecs
 - ELD notifier implenetation for Intel HDMI/DP
 
 USB-audio:
 - Refactor runtime PM code to make lockdep happier
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Merge tag 'sound-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There are little changes in core part, but lots of development are
  found in drivers, especially ASoC.  The diffstat shows regmap-related
  changes for a slight API additions / changes, and that's all.

  Looking at the code size statistics, the most significant addition is
  for Intel Skylake.  (Note that SKL support is still underway, the
  codec driver is missing.) Also STI controller driver is a major
  addition as well as a few new codec drivers.

  In HD-audio side, there are fewer changes than the past.  The
  noticeable change is the support of ELD notification from i915
  graphics driver.  Thus this pull request carries a few changes in
  drm/i915.

  Other than that, USB-audio got a rewrite of runtime PM code.  It was
  initiated by lockdep warning, but resulted in a good cleanup in the
  end.

  Below are the highlights:

  Common:
   - Factoring out of AC'97 reset code from ASoC into the core helper
   - A few regmap API extensions (in case it's not pulled yet)

  ASoC:
   - New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
     RT298 and ST STI controllers
   - Machine drivers for Rockchip systems with MAX98090 and RT5645 and
     RT5650
   - Initial driver support for Intel Skylake devices
   - Lots of rsnd cleanup and enhancements
   - A few DAPM fixes and cleanups
   - A large number of cleanups in various drivers (conversion and
     standardized to regmap, component) mostly by Lars-Peter and Axel

  HD-audio:
   - Extended HD-audio core for Intel Skylake controller support
   - Quirks for Dell headsets, Alienware 15
   - Clean up of pin-based quirk tables for Realtek codecs
   - ELD notifier implenetation for Intel HDMI/DP

  USB-audio:
   - Refactor runtime PM code to make lockdep happier"

* tag 'sound-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
  drm/i915: Add locks around audio component bind/unbind
  drm/i915: Drop port_mst_index parameter from pin/eld callback
  ALSA: hda - Fix missing inline for dummy snd_hdac_set_codec_wakeup()
  ALSA: hda - Wake the codec up on pin/ELD notify events
  ALSA: hda - allow codecs to access the i915 pin/ELD callback
  drm/i915: Call audio pin/ELD notify function
  drm/i915: Add audio pin sense / ELD callback
  ASoC: zx296702-i2s: Fix resource leak when unload module
  ASoC: sti_uniperif: Ensure component is unregistered when unload module
  ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource
  ASoC: sh: dma-sh7760: Convert to devm_snd_soc_register_platform
  ASoC: spear_pcm: Use devm_snd_dmaengine_pcm_register to fix resource leak
  ALSA: fireworks/bebob/dice/oxfw: fix substreams counting at vmalloc failure
  ASoC: Clean up docbook warnings
  ASoC: txx9: Convert to devm_snd_soc_register_platform
  ASoC: pxa: Convert to devm_snd_soc_register_platform
  ASoC: nuc900: Convert to devm_snd_soc_register_platform
  ASoC: blackfin: Convert to devm_snd_soc_register_platform
  ASoC: au1x: Convert to devm_snd_soc_register_platform
  ASoC: qcom: Constify asoc_qcom_lpass_cpu_dai_ops
  ...
2015-09-04 11:46:02 -07:00
Luis de Bethencourt 86bfbc1566 ALSA: sparc: amd7930: Fix module autoload for OF platform driver
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.

Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-04 12:11:44 +02:00
Woodrow Shen 1adecc6755 ALSA: hda - Add some FIXUP quirks for white noise on Dell laptop.
Dell laptop has a series model to use the same codec but different subsystem ID.
At the same time they happens the white noise by login screen and headphone;
for fixing them together, I only can add these IDs to FIXUP function ALC292_FIXUP_DISABLE_AAMIX,
then try to solve such the similar issues.

Codec: Realtek ALC3235
Vendor Id: 0x10ec0293
Subsystem Id: 0x102806dd
Subsystem Id: 0x102806df
Subsystem Id: 0x102806e0

Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1492132
Signed-off-by: Woodrow Shen <woodrow.shen@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-04 09:13:47 +02:00
David Henningsson f0675d4a8e drm/i915: Drop port_mst_index parameter from pin/eld callback
The port_mst_index parameter was reserved for future use, but
maintainers prefer to add it later when it is actually used.

[Note: this is an update patch to commit [51e1d83cab99: drm/i915: Call
 audio pin/ELD notify function] where I mistakenly applied the older
 version.  Jani and Daniel's review tags were to the latest version,
 so I add them below, too -- tiwai]

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Reviewed-by: Jani Nikula <jani.nikula@intel.com>
Acked-by: Daniel Vetter <daniel@ffwll.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-03 12:09:03 +02:00
David Henningsson 25adc137c5 ALSA: hda - Wake the codec up on pin/ELD notify events
Whenever there is an event from the i915 driver, wake the codec
and recheck plug/unplug + ELD status.

This fixes the issue with lost unsol events in power save mode,
the codec and controller can now sleep in D3 and still know when
the HDMI monitor has been connected.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-02 12:24:40 +02:00
David Henningsson 45c053df5b ALSA: hda - allow codecs to access the i915 pin/ELD callback
This lets the interested codec be notified when an i915 pin/ELD
event happens.

[tiwai: Fixed a trivial build error for CONFIG_SND_HDA_I915=n]

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-09-02 12:23:55 +02:00
Linus Torvalds 089b669506 Merge branch 'for-next' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
Pull trivial tree updates from Jiri Kosina:
 "The usual stuff from trivial tree for 4.3 (kerneldoc updates, printk()
  fixes, Documentation and MAINTAINERS updates)"

* 'for-next' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (28 commits)
  MAINTAINERS: update my e-mail address
  mod_devicetable: add space before */
  scsi: a100u2w: trivial typo in printk
  i2c: Fix typo in i2c-bfin-twi.c
  treewide: fix typos in comment blocks
  Doc: fix trivial typo in SubmittingPatches
  proportions: Spelling s/consitent/consistent/
  dm: Spelling s/consitent/consistent/
  aic7xxx: Fix typo in error message
  pcmcia: Fix typo in locking documentation
  scsi/arcmsr: Fix typos in error log
  drm/nouveau/gr: Fix typo in nv10.c
  [SCSI] Fix printk typos in drivers/scsi
  staging: comedi: Grammar s/Enable support a/Enable support for a/
  Btrfs: Spelling s/consitent/consistent/
  README: GTK+ is a acronym
  ASoC: omap: Fix typo in config option description
  mm: tlb.c: Fix error message
  ntfs: super.c: Fix error log
  fix typo in Documentation/SubmittingPatches
  ...
2015-09-01 18:46:42 -07:00
Sudip Mukherjee 51d2eeef1d ASoC: wm0010: fix memory leak
We were aborting if the kzalloc of img_swap fails but without freeing the
already allocated out. Similarly we were aborting if spi_sync fails
without releasing out and img_swap.

Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-01 14:10:50 +01:00
Anatol Pomozov d76f41982f ASoC: Document snd-soc-dummy-dai purpose
Signed-off-by: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-31 16:40:26 +01:00
Axel Lin 887e83e32d ASoC: mediatek: Convert to devm_snd_soc_register_card
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-31 16:11:32 +01:00
Maciej S. Szmigiero 6bd3c6f75e ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device
In case of unknown DT compatible device the ASRC OF node
possibly acquired earlier by of_parse_phandle() has
to be put before returning from probe method.

Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-31 16:08:50 +01:00
Takashi Iwai 08ceab9d87 ASoC: Updates for v4.3
Not many updates to the core here, but an awful lot of driver updates
 this time round:
 
  - Factoring out of AC'97 reset code into the core
  - New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
    RT298 and ST STI controllers.
  - Machine drivers for Rockchip systems with MAX98090 and RT5645 and
    RT5650.
  - Initial driver support for Intel Skylake devices.
  - A large number of cleanups for Lars-Peter Clausen and Axel Lin.
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Merge tag 'asoc-v4.2-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Updates for v4.3

Not many updates to the core here, but an awful lot of driver updates
this time round:

 - Factoring out of AC'97 reset code into the core
 - New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
   RT298 and ST STI controllers.
 - Machine drivers for Rockchip systems with MAX98090 and RT5645 and
   RT5650.
 - Initial driver support for Intel Skylake devices.
 - A large number of cleanups for Lars-Peter Clausen and Axel Lin.
2015-08-31 16:25:22 +02:00
Mark Brown 91931320cf Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/xtensa' and 'asoc/topic/zx296702' into asoc-next 2015-08-30 15:59:18 +01:00
Mark Brown 02dc14d66e Merge remote-tracking branches 'asoc/topic/wm8741', 'asoc/topic/wm8753', 'asoc/topic/wm8904', 'asoc/topic/wm8960' and 'asoc/topic/wm8983' into asoc-next 2015-08-30 15:58:56 +01:00
Mark Brown 384ea7936c Merge remote-tracking branches 'asoc/topic/wm0010', 'asoc/topic/wm5100', 'asoc/topic/wm5110', 'asoc/topic/wm8004' and 'asoc/topic/wm8731' into asoc-next 2015-08-30 15:58:36 +01:00
Mark Brown c009d370dd Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/uda134x', 'asoc/topic/ux500' and 'asoc/topic/width' into asoc-next 2015-08-30 15:58:18 +01:00
Mark Brown 428157c1e8 Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tas5086', 'asoc/topic/tegra', 'asoc/topic/tlv' and 'asoc/topic/topology' into asoc-next 2015-08-30 15:57:34 +01:00
Mark Brown 0a5ff07757 Merge remote-tracking branches 'asoc/topic/ssm2518', 'asoc/topic/sta529', 'asoc/topic/sti' and 'asoc/topic/sti-sas' into asoc-next 2015-08-30 15:57:13 +01:00
Mark Brown cfed47d7b5 Merge remote-tracking branches 'asoc/topic/rt5677', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf-codec' and 'asoc/topic/spear' into asoc-next 2015-08-30 15:56:44 +01:00
Mark Brown a8f1d97bfa Merge remote-tracking branches 'asoc/topic/rt298', 'asoc/topic/rt5640', 'asoc/topic/rt5651' and 'asoc/topic/rt5670' into asoc-next 2015-08-30 15:56:27 +01:00
Mark Brown 532161e6cc Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/reg-default', 'asoc/topic/rl6231', 'asoc/topic/rockchip' and 'asoc/topic/rt286' into asoc-next 2015-08-30 15:55:54 +01:00
Mark Brown d781c23e9b Merge remote-tracking branches 'asoc/topic/pm-ops', 'asoc/topic/pxa' and 'asoc/topic/qcom' into asoc-next 2015-08-30 15:55:41 +01:00
Mark Brown acb47ad502 Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/mtk', 'asoc/topic/nuc900', 'asoc/topic/of-name' and 'asoc/topic/omap' into asoc-next 2015-08-30 15:55:21 +01:00
Mark Brown 1d1ed2c23e Merge remote-tracking branches 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/max98357a', 'asoc/topic/max9877' and 'asoc/topic/max98925' into asoc-next 2015-08-30 15:54:57 +01:00
Mark Brown 373e515d65 Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/kirkwood', 'asoc/topic/lm49453', 'asoc/topic/max9768' and 'asoc/topic/max98088' into asoc-next 2015-08-30 15:54:38 +01:00
Mark Brown 246647ee3e Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/fsl-ssi', 'asoc/topic/gtm601', 'asoc/topic/ics43432' and 'asoc/topic/ids' into asoc-next 2015-08-30 15:54:15 +01:00
Mark Brown bb85d37358 Merge remote-tracking branches 'asoc/topic/fsi', 'asoc/topic/fsl', 'asoc/topic/fsl-asrc', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-sai' into asoc-next 2015-08-30 15:53:56 +01:00
Mark Brown 7c0031360b Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/davinci-vcif', 'asoc/topic/doc' and 'asoc/topic/dpcm' into asoc-next 2015-08-30 15:53:39 +01:00
Mark Brown 399962239c Merge remote-tracking branches 'asoc/topic/cs42l56', 'asoc/topic/cs42l73', 'asoc/topic/cs4349' and 'asoc/topic/da732x' into asoc-next 2015-08-30 15:53:22 +01:00
Mark Brown b18fec9fe4 Merge remote-tracking branches 'asoc/topic/const', 'asoc/topic/cs35l32', 'asoc/topic/cs4265' and 'asoc/topic/cs42l52' into asoc-next 2015-08-30 15:52:59 +01:00
Mark Brown 48997b9c03 Merge remote-tracking branches 'asoc/topic/au1x', 'asoc/topic/bcm2835', 'asoc/topic/blackfin' and 'asoc/topic/card' into asoc-next 2015-08-30 15:52:41 +01:00
Mark Brown 21af109438 Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ac97', 'asoc/topic/ak4542', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next 2015-08-30 15:52:21 +01:00