Commit Graph

30451 Commits

Author SHA1 Message Date
Mark Brown b0a39d356a
ASoC: wcd9335: Fix build due to CLASS-H Controller support
This reverts commit c8cb5f775c (ASoC: vert "ASoC: wcd9335: add
CLASS-H Controller support) due to missing dependencies.

Signed-off-by: Mark Brown <broonie@kernel.org>
2018-08-01 14:18:50 +01:00
Colin Ian King d36455a38e ALSA: usb-audio: remove redundant pointer 'urb'
Pointer 'urb' is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'urb' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-08-01 14:00:32 +02:00
Colin Ian King 0d00085b90 ALSA: sonicvibes: remove redundant pointer 'dir'
Pointer 'dir' is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'dir' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-08-01 14:00:13 +02:00
Colin Ian King 3b0cbc7812 ALSA: ens137x: remove redundant array pcm_devs
The array pcm_devs is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'pcm_devs' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-08-01 13:59:58 +02:00
Colin Ian King de42b4b96e ALSA: emu10k1: remove redundant variable attn
Variable attn is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'attn' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-08-01 13:59:39 +02:00
Colin Ian King 45bf41005a ALSA: cs5535audio: remove redundant pointer 'dma'
Pointer 'dma' is being assigned but is never used hence it is
redundant and can be removed.

Cleans up two clang warnings:
warning: variable 'dma' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-08-01 13:59:29 +02:00
Colin Ian King 96963dedd0 ALSA: asihpi: remove redundant variable max_streams
Variable max_streams is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'max_streams' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-08-01 13:59:12 +02:00
Colin Ian King 18127744cf
ASoC: stm32: remove redundant pointers 'priv' and 'rtd'
Pointer 'priv' is assigned and not used, removing this allows
the removal of pointer 'rtd'.

Cleans up clang warning:
warning: variable 'priv' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-08-01 12:16:26 +01:00
Colin Ian King d101f9b96e
ASoC: nau8540: remove redundant variable osrate
Variable osrate is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'osrate' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-08-01 12:16:22 +01:00
Rohit kumar 6b1687bf76
ASoC: qcom: add sdm845 sound card support
This patch adds sdm845 audio machine driver support.

Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-08-01 12:00:25 +01:00
Rohit kumar c25e295cd7
ASoC: qcom: Add support to parse common audio device nodes
This adds support to parse cpu, platform and codec
device nodes and add them in dai-links. Also, update
apq8096 machine driver to use the common API.

Acked-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-08-01 12:00:21 +01:00
Mark Brown a0b5031582
Merge branch 'topic/drm_audio_component' of https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound into asoc-4.19 2018-08-01 10:32:05 +01:00
Srinivas Kandagatla c8cb5f775c
ASoC: wcd9335: add CLASS-H Controller support
CLASS-H controller/Amplifier is common accorss Qualcomm WCD codec series.
This patchset adds basic CLASS-H controller apis for WCD codecs after
wcd9335 to use.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-31 18:17:37 +01:00
Srinivas Kandagatla e57d4ca882
ASoC: wcd9335: add support to wcd9335 codec
Qualcomm WCD9335 Codec is a standalone Hi-Fi audio codec IC,
It supports both I2S/I2C and SLIMbus audio interfaces.
On slimbus interface it supports two data lanes; 16 Tx ports
and 8 Rx ports. It has Seven DACs and nine dedicated interpolators,
Seven (six audio ADCs, and one VBAT ADC), Multibutton headset
control (MBHC), Active noise cancellation and Sidetone paths
and processing.

This patchset adds very basic support for playback and capture
via the 9 interpolators and ADC respectively.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-31 18:17:32 +01:00
Jorge Sanjuan a0a4959eb4 ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks
Make use of UAC3 Power Domains associated to an Audio Streaming
path within the PCM's logic. This means, when there is no audio
being transferred (pcm is closed), the host will set the Power Domain
associated to that substream to state D1. When audio is being transferred
(from hw_params onwards), the Power Domain will be set to D0 state.

This is the way the host lets the device know which Terminal
is going to be actively used and it is for the device to
manage its own internal resources on that UAC3 Power Domain.

Note the resume method now sets the Power Domain to D1 state as
resuming the device doesn't mean audio streaming will occur.

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-31 15:01:45 +02:00
Jorge Sanjuan 3f59aa11c6 ALSA: usb-audio: Add UAC3 Power Domains to suspend/resume
Set the UAC3 Power Domain state for an Audio Streaming interface
to D2 state before suspending the device (usb_driver callback).
This lets the device know there is no intention to use any of the
Units in the Audio Function and that the host is not going to
even listen for wake-up events (interrupts) on the units.

When the usb_driver gets resumed, the state D0 (fully powered) will
be set. This ties up the UAC3 Power Domains to the runtime PM.

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-31 15:01:36 +02:00
Jorge Sanjuan 7edf3b5e6a ALSA: usb-audio: AudioStreaming Power Domain parsing
Power Domains in the UAC3 spec are mainly intended to be
associated to an Input or Output Terminal so the host
changes the power state of the entire capture or playback
path within the topology.

This patch adds support for finding Power Domains associated
to an Audio Streaming Interface (bTerminalLink) and adds a
reference to them in the usb audio substreams (snd_usb_substream).

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-31 15:01:30 +02:00
Jorge Sanjuan 11785ef532 ALSA: usb-audio: Initial Power Domain support
Thee USB Audio Class 3 (UAC3) introduces Power Domains as a new
feature to let a host turn individual parts of an audio function
to different power states via USB requests. This lets the device
get to know a bit amore about what the host is up to in order to
optimize power consumption efficiently.

The Power Domains are optional for UAC3 configuration but all
UAC3 devices shall include at least one BADD configuration where
the support for Power Domains is compulsory.

This patch adds a set of features/helpers to parse these power
domains and change their status.

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-31 15:01:22 +02:00
Takashi Iwai 89b4ab213f ALSA: seq: virmidi: Use READ_ONCE/WRITE_ONCE() macros
The trigger flag in vmidi object can be referred in different contexts
concurrently, hence it's better to be put with READ_ONCE() and
WRITE_ONCE() macros to assure the accesses.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-30 14:52:30 +02:00
Takashi Iwai f7debfe540 ALSA: seq: virmidi: Offload the output event processing
The virmidi sequencer stuff tries to translate the rawmidi bytes to
sequencer events and deliver the packets at trigger callback.  The
amount of the whole process of these translations and deliveries
depends on the incoming rawmidi bytes, and we have no limit for that;
this was the cause of a CPU soft lockup that had been reported and
fixed recently.

Although we've fixed the soft lockup by putting the temporary unlock
and cond_resched(), it's rather a quick band aid.  In this patch,
meanwhile, the event parsing and delivery process is offloaded to a
dedicated work, and the trigger callback just kicks it off.  It has
three merits, at least:

- The processing is always done in a sleepable context, which can
  assure the event delivery with non-atomic flag without hackish
  is_atomic() usage.

- Other relevant codes can be simplified, reducing the lines

- It makes me happier

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-30 14:51:51 +02:00
Katsuhiro Suzuki d8504acca7
ASoC: uniphier: change functions to static
This patch changes some functions that are not used by other objects
to static.

Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro@socionext.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:32 +01:00
Katsuhiro Suzuki 8fc9983db1
ASoC: uniphier: add support for multichannel output
This patch adds multichannel PCM output support for LD11/LD20.
Currently driver tested and supported only 2ch, 6ch, and 8ch.

Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro@socionext.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:31 +01:00
Gustavo A. R. Silva ae1c696a48
ASoC: sirf: Fix potential NULL pointer dereference
There is a potential execution path in which function
platform_get_resource() returns NULL. If this happens,
we will end up having a NULL pointer dereference.

Fix this by replacing devm_ioremap with devm_ioremap_resource,
which has the NULL check and the memory region request.

This code was detected with the help of Coccinelle.

Cc: stable@vger.kernel.org
Fixes: 2bd8d1d5cf ("ASoC: sirf: Add audio usp interface driver")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:30 +01:00
Alexey Khoroshilov 4321723648
ASoC: tegra_alc5632: fix device_node refcounting
tegra_alc5632_probe() increments reference count of device nodes
with of_parse_phandle(), but there is no code decrementing them
in the driver.

The patch adds of_node_put() to tegra_alc5632_remove() and
to error handling paths in the probe.

Found by Linux Driver Verification project (linuxtesting.org).

Signed-off-by: Alexey Khoroshilov <khoroshilov@ispras.ru>
Acked-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:29 +01:00
Kuninori Morimoto 7464d3faf6
ASoC: sh: Kconfig: convert to SPDX identifiers
By default all files without license information are under the default
license of the kernel, which is GPL version 2.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:29 +01:00
Kuninori Morimoto e028937c77
ASoC: ak4613: convert to SPDX identifiers
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:28 +01:00
Kuninori Morimoto c0ca5604d4
ASoC: da7210: convert to SPDX identifiers
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:27 +01:00
Kuninori Morimoto 7a968dc66a
ASoC: ak4554: convert to SPDX identifiers
As original license mentioned, it is GPL-2.0 in SPDX.
Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL".
See ${LINUX}/include/linux/module.h

	"GPL"           [GNU Public License v2 or later]
	"GPL v2"        [GNU Public License v2]

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:26 +01:00
Kuninori Morimoto fe209b9718
ASoC: ak4642: convert to SPDX identifiers
As original license mentioned, it is GPL-2.0 in SPDX.
Then, MODULE_LICENSE() should be "GPL v2" instead of "GPL".
See ${LINUX}/include/linux/module.h

	"GPL"           [GNU Public License v2 or later]
	"GPL v2"        [GNU Public License v2]

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 12:02:25 +01:00
Ladislav Michl c9c9780d8f
ASoC: wm8988: fix typo in rate constraints
Remove duplicated entry and add missing zero in rate constraints.

Signed-off-by: Ladislav Michl <ladis@linux-mips.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 11:18:32 +01:00
Edward Cragg 279fef50b6
ASoC: tegra: i2s: Fix typo/broken macro
Fix typo in macro TEGRA30_I2S_SLOT_CTRL_TOTAL_SLOTS_MASK.

Signed-off-by: Edward Cragg <edward.cragg@codethink.co.uk>
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-30 11:18:28 +01:00
Takashi Iwai 16c796e8fa Merge branch 'for-linus' into topic/virmidi
Pull the latest ALSA sequencer fixes for the further development of
virmidi.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-29 22:39:29 +02:00
Takashi Iwai f69548ffaf ALSA: hda/hdmi: Use single mutex unlock in error paths
Instead of calling mutex_unlock() at each error path multiple times,
take the standard goto-and-a-single-unlock approach.  This will
simplify the code and make easier to find the unbalanced mutex locks.

No functional changes, but only the code readability improvement as a
preliminary work for further changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-29 09:28:12 +02:00
Park Ju Hyung f59cf9a055 ALSA: hda - Sleep for 10ms after entering D3 on Conexant codecs
On rare occasions, we are still noticing that the internal speaker
spitting out spurious noises even after adding the problematic codec
to the list.

Adding a 10ms artificial delay before rebooting fixes the issue entirely.

Patch for Realtek codecs also adds the same amount of delay after
entering D3.

Signed-off-by: Park Ju Hyung <qkrwngud825@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-28 18:57:56 +02:00
Park Ju Hyung d77a4b4a5b ALSA: hda - Turn CX8200 into D3 as well upon reboot
As an equivalent codec with CX20724,
CX8200 is also subject to the reboot bug.

Late 2017 and 2018 LG Gram and some HP Spectre laptops are known victims
to this issue, causing extremely loud noises upon reboot.

Now that we know that this bug is subject to multiple codecs,
fix the comment as well.

Signed-off-by: Park Ju Hyung <qkrwngud825@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-28 18:57:33 +02:00
Jia-Ju Bai fad56c895f ALSA: ctxfi: cthw20k2: Replace mdelay() with msleep() and usleep_range()
hw_pll_init(), hw_dac_stop(), hw_dac_start() and hw_adc_init()
are never called in atomic context.
They call mdelay() to busily wait, which is not necessary.
mdelay() can be replaced with msleep().

This is found by a static analysis tool named DCNS written by myself.

Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 11:49:16 +02:00
Jia-Ju Bai 08fd8325d9 ALSA:: ctxfi: cthw20k1: Replace mdelay() with msleep()
hw_pll_init(), hw_reset_dac() and hw_card_init() are never
called in atomic context.
They calls mdelay() to busily wait, which is not necessary.
mdelay() can be replaced with msleep().

This is found by a static analysis tool named DCNS written by myself.

Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 11:48:50 +02:00
Jia-Ju Bai df3f0347fd ALSA: usb-audio: quirks: Replace mdelay() with msleep() and usleep_range()
snd_usb_select_mode_quirk(), snd_usb_set_interface_quirk() and
snd_usb_ctl_msg_quirk() are never called in atomic context.
They call mdelay() to busily wait, which is not necessary.
mdelay() can be replaced with msleep() and usleep_range().

This is found by a static analysis tool named DCNS written by myself.

Signed-off-by: Jia-Ju Bai <baijiaju1990@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 11:48:07 +02:00
Takashi Iwai 13e9a3edb4 ALSA: sb: Proper endian notations
The data types defined in SB CSP driver code are all in little-endian,
hence the proper type like __le32 should be used.

Spotted by sparse, a warning like:
  sound/isa/sb/sb16_csp.c:330:14: warning: cast to restricted __le32

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:14 +02:00
Takashi Iwai 7e49aadf64 ALSA: atiixp_modem: Proper endian notations
The DMA address table in atiixp modem driver is in little-endian,
hence we should define it with __le32 properly.

Spotted by sparse, a warning like:
  sound/pci/atiixp_modem.c:360:28: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:13 +02:00
Takashi Iwai c44a81a40a ALSA: atiixp: Proper endian notations
The DMA address table in atiixp driver is in little-endian, hence we should define it with __le32 properly.

Spotted by sparse, a warning like:
  sound/pci/atiixp.c:393:28: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:12 +02:00
Takashi Iwai 58578d1894 ALSA: bt87x: Proper endian notations
The RISC data in bt87x is in little-endian, hence we should define it
with __le32 properly.

Spotted by sparse, a warning like:
  sound/pci/bt87x.c:240:17: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:11 +02:00
Takashi Iwai 2a833a02a1 ALSA: echoaudio: Proper endian notations
Many data fields defined in echoaudio drivers are in little-endian,
hence they should be defined with __le16 or __le32.  This makes it
easier to catch the forgotten conversions.

Spotted by sparse, a warning like:
  sound/pci/echoaudio/echoaudio_dsp.c:990:36: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:09 +02:00
Takashi Iwai 8c0ab942e0 ALSA: maestro3: Proper endian notations
The ASSP data passed to maestro3 driver is in little-endian format,
hence the data pointer should be with __le16.

Spotted by sparse, warnings like:
  sound/pci/maestro3.c:2128:35: warning: cast to restricted __le16

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:08 +02:00
Takashi Iwai 7752a7de25 ALSA: intel8x0m: Proper endian notations
The BD address tables in intel8x0m driver are in little-endian, hence
they should be represented as __le32 instead u32.

Spotted by sparse, warnings like:
  sound/pci/intel8x0m.c:406:40: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:07 +02:00
Takashi Iwai 3c164e2ce6 ALSA: intel8x0: Proper endian notations
The BD address tables in intel8x0 driver are in little-endian, hence
they should be represented as __le32 instead u32.

Spotted by sparse, warnings like:
  sound/pci/intel8x0.c:688:40: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:06 +02:00
Takashi Iwai 0d9a26fc74 ALSA: lola: Proper endian notations
The BDL entries in lola driver are little-endian while we code them as
u32.  This leads to sparse warnings like:
  sound/pci/lola/lola.c:105:40: warning: incorrect type in assignment (different base types)
  sound/pci/lola/lola.c:105:40:    expected unsigned int [unsigned] [usertype] <noident>
  sound/pci/lola/lola.c:105:40:    got restricted __le32 [usertype] <noident>

This patch fixes the declarations to the proper __le32 type.

Also, there was a typo in the original code, where __user was used
that was intended as __iomem.  This was caused also by sparse:
  sound/pci/lola/lola_mixer.c:132:27: warning: incorrect type in assignment (different address spaces)
Fixed in this patch as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:05 +02:00
Takashi Iwai 0e7ca66a97 ALSA: mixart: Proper endian notations
The miXart driver deals with big-endian values as raw data, while it
declares most of variables as u32.  This leads to sparse warnings like
  sound/pci/mixart/mixart.c:1203:23: warning: cast to restricted __be32

Fix them by properly defining the structs and add the explicit cast to
macros.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:03 +02:00
Takashi Iwai be05e3de3a ALSA: riptide: Properly endian notations
The SG descriptor of Riptide contains the little-endian values, hence
we need to define with __le32 properly.  This fixes sparse warnings
like:
  sound/pci/riptide/riptide.c:1112:40: warning: cast to restricted __le32

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:02 +02:00
Takashi Iwai 7362b0fca5 ALSA: hda: Proper endian notations for BDL pointers
The BDL pointer used in snd_hdac_dsp_prepare() should be declared as
__le32, as warned by sparse:
  sound/hda/hdac_stream.c:655:47: warning: incorrect type in argument 4 (different base types)
  sound/hda/hdac_stream.c:655:47:    expected restricted __le32 [usertype] **bdlp
  sound/hda/hdac_stream.c:655:47:    got unsigned int [usertype] **<noident>

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:01 +02:00
Takashi Iwai 752089fea3 ALSA: trident: Proper endian notations
The TLB entries in Trident driver are represented in little-endian,
hence they should be declared as __le32.

This patch fixes the sparse warnings like:
  sound/pci/trident/trident_memory.c:226:17: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:06:00 +02:00
Takashi Iwai d3c637632d ALSA: ymfpci: Proper endian notations
The bank values are all little-endians, so they should be defined with
__le32.  This fixes lots of sparse warnings like:
  sound/pci/ymfpci/ymfpci_main.c:315:23: warning: cast to restricted __le32
  sound/pci/ymfpci/ymfpci_main.c:342:32: warning: incorrect type in assignment (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:58 +02:00
Takashi Iwai 3ac14b3960 ALSA: xen: Use standard pcm_format_to_bits() for ALSA format bits
The open codes with the bit shift in xen_snd_front_alsa.c give sparse
warnings as the PCM format type is with __bitwise.
There is already a standard macro to get the format bits, so let's use
it instead.

This fixes sparse warnings like:
  sound/xen/xen_snd_front_alsa.c:191:47: warning: restricted snd_pcm_format_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:34 +02:00
Takashi Iwai e5d3765b6c ALSA: sb: Fix sparse warning wrt PCM format type
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly.  Instead of an ugly cast, declare the function
argument of snd_sb_csp_autoload() with the proper snd_pcm_format_t
type.

This fixes the sparse warnings like:
  sound/isa/sb/sb16_csp.c:743:22: warning: restricted snd_pcm_format_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:33 +02:00
Takashi Iwai 55ff2d1ea5 ALSA: sb: Fix PCM format bit calculation
The PCM format type in snd_pcm_format_t can't be treated as integer
implicitly since it's with __bitwise.  We have already a helper
function to get the bit index of the given type, and use it in each
place instead.

This fixes sparse warnings like:
  sound/isa/sb/sb16_main.c:61:44: warning: restricted snd_pcm_format_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:32 +02:00
Takashi Iwai 6be9a60efb ALSA: wss: Fix sparse warning wrt PCM format type
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly.  Instead of an ugly cast, declare the function
argument of snd_wss_get_format() with the proper snd_pcm_format_t
type.

This fixes the sparse warnings like:
  sound/isa/wss/wss_lib.c:551:14: warning: restricted snd_pcm_format_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:31 +02:00
Takashi Iwai a91a0e7749 ALSA: asihpi: Fix PCM format notations
asihpi driver treats -1 as an own invalid PCM format, but this needs
a proper cast with __force prefix since PCM format type is __bitwise.
Define a constant with the proper type and use it allover.

This fixes sparse warnings like:
  sound/pci/asihpi/asihpi.c:315:9: warning: incorrect type in initializer (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:30 +02:00
Takashi Iwai 10d3d91e3b ALSA: au88x0: Fix sparse warning wrt PCM format type
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly.  Instead of an ugly cast, declare the function
argument of vortex_alsafmt_aspfmt() with the proper snd_pcm_format_t
type.

This fixes the sparse warning like:
  sound/pci/au88x0/au88x0_core.c:2778:14: warning: restricted snd_pcm_format_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:28 +02:00
Takashi Iwai d63f33d3f0 ALSA: ad1816a: Fix sparse warning wrt PCM format type
The PCM format type is with __bitwise, and it can't be converted from
integer implicitly.  Instead of an ugly cast, declare the function
argument of snd_ad1816a_get_format() with the proper snd_pcm_format_t
type.

This fixes the sparse warning like:
  sound/isa/ad1816a/ad1816a_lib.c:93:14: warning: restricted snd_pcm_format_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:27 +02:00
Takashi Iwai f8b6c0cfbd ALSA: pcm: Fix sparse warning wrt PCM format type
The PCM format type is with __bitwise, hence it needs the explicit
cast with __force.  It's ugly, but there is a reason for that cost...

This fixes the sparse warning:
  sound/core/oss/pcm_oss.c:1854:55: warning: incorrect type in argument 1 (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:27 +02:00
Takashi Iwai 94dfee0c1a ALSA: riptide: Fix PCM format type conversion
The PCM format type is with __bitwise, hence it needs to be explicitly
declared as snd_pcm_format_t, as warned by sparse:
  sound/pci/riptide/riptide.c:1028:34: warning: incorrect type in argument 1 (different base types)
  sound/pci/riptide/riptide.c:1028:34:    expected restricted snd_pcm_format_t [usertype] format
  sound/pci/riptide/riptide.c:1028:34:    got unsigned char [unsigned] format

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:26 +02:00
Takashi Iwai a6ea5fe95a ALSA: hda: Fix implicit PCM format type conversion
The PCM format type is defined with __bitwise, hence it can't be
passed as integer but needs an explicit cast.  In this patch, instead
of the messy cast flood, define the format argument of
snd_hdac_calc_stream_format() to be the proper snd_pcm_format_t type.

This fixes sparse warnings like:
  sound/hda/hdac_device.c:760:38: warning: incorrect type in argument 1 (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 09:05:24 +02:00
Takashi Iwai 50e9ffb199 ALSA: virmidi: Fix too long output trigger loop
The virmidi output trigger tries to parse the all available bytes and
process sequencer events as much as possible.  In a normal situation,
this is supposed to be relatively short, but a program may give a huge
buffer and it'll take a long time in a single spin lock, which may
eventually lead to a soft lockup.

This patch simply adds a workaround, a cond_resched() call in the loop
if applicable.  A better solution would be to move the event processor
into a work, but let's put a duct-tape quickly at first.

Reported-and-tested-by: Dae R. Jeong <threeearcat@gmail.com>
Reported-by: syzbot+619d9f40141d826b097e@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-27 08:59:25 +02:00
Takashi Iwai 8adf3df415
ASoC: dmaengine: Use standard pcm_format_to_bits() macro
The conversion from PCM format type to bits needs an explicit cast,
and it'll be uglier.  Since we have a standard macro for that, let's
use it instead.

This patch fixes the sparse warning:
  sound/soc/soc-generic-dmaengine-pcm.c:200:63: warning: restricted snd_pcm_format_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 17:09:09 +01:00
Takashi Iwai 79b8a50813
ASoC: pcm186x: Declare PCM format with snd_pcm_format_t
The PCM format type is with __bitwise, so we should use the dedicated
snd_pcm_format_t instead of int.

This fixes the sparse warning like:
  sound/soc/codecs/pcm186x.c:268:44: warning: incorrect type in initializer (different base types)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 17:09:08 +01:00
Takashi Iwai ebc22af0c9
ASoC: fsl: Use snd_mask_set_format()
Use the new helper function snd_mask_set_format() for avoiding the
ugly cast with __force prefix.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 17:09:07 +01:00
Takashi Iwai b5453e8ca3
ASoC: intel: Fix snd_pcm_format_t handling
As sparse warns, the PCM format type can't be dealt as integer as
found in Intel SST driver codes.

Fix them in the following two ways:

- The open code with snd_mask_set() and params->masks reference is
  replaced with params_set_format()

- The rest codes with snd_mask_set(fmt, SNDRV_PCM_FORMAT_XXX) are
  replaced with the new helper, snd_mask_set_format().

Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 17:09:06 +01:00
Takashi Iwai 3ba66feb59
ASoC: dapm: Use int for format bit position
fmt in snd_soc_dai_link_event() contains the format bit position, not
the format bit itself.  Hence it can be a simple integer instead of
the explicit u64.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 15:48:21 +01:00
Takashi Iwai 40d1299f87
ASoC: dmaengine: Fix missing __user prefix in copy_user callback
It seems that __user prefix was forgotten to be added to
dmaengine_copy_user callback while we refactored the user-copy PCM
core.

This patch adds the missing prefix, remove the superfluous cast, and
add the needed cast (__force is needed for downgrading from user
pointer to kernel pointer), too.

Spotted by a sparse warning like:
  sound/soc/soc-generic-dmaengine-pcm.c:397:27: warning: incorrect type in initializer (incompatible argument 4 (different address spaces))

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 15:48:20 +01:00
Takashi Iwai c889a45d22
ASoC: zte: Fix incorrect PCM format bit usages
zx-tdm driver sets the DAI driver definitions with the format bits
wrongly set with SNDRV_PCM_FORMAT_*, instead of SNDRV_PCM_FMTBIT_*.

This patch corrects the definitions.

Spotted by a sparse warning:
  sound/soc/zte/zx-tdm.c:363:35: warning: restricted snd_pcm_format_t degrades to integer

Fixes: 870e0ddc43 ("ASoC: zx-tdm: add zte's tdm controller driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 15:48:19 +01:00
Jerome Brunet 435857e015
ASoC: meson: align axg card driver with DT bindings documentation
Drop amlogic prefix in front of the generic DT properties and change
property "name" to "model".

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 15:45:44 +01:00
Jerome Brunet 036e4963bf
ASoC: meson: use IRQ_RETVAL in the fifo irq handler
Use IRQ_RETVAL instead of the open coded ternary operation.

Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 15:45:42 +01:00
Akshu Agrawal 7699676081
ASoC: AMD: Fix build warning
Fixes
sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_probe':
sound/soc/amd/acp-da7219-max98357a.c:367:3: warning: 'ret' may
be used uninitialized in this function [-Wmaybe-uninitialized]
   dev_err(&pdev->dev, "Failed to register regulator: %d\n",
ret);

Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 14:07:55 +01:00
Akshu Agrawal c183fec10a
ASoC: AMD: Add a fix voltage regulator for DA7219 and ADAU7002
DA7219 for our platform need to be configured for 1.8V.
Hence, we add a volatge regulator with supplies
of 1.8V in the machine driver.

Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-26 14:07:47 +01:00
Takashi Iwai d6b340d7cb ALSA: trident: Suppress gcc string warning
The meddlesome gcc warns about the possible shortname string in
trident driver code:
  sound/pci/trident/trident.c: In function ‘snd_trident_probe’:
  sound/pci/trident/trident.c:126:2: warning: ‘strcat’ accessing 17 or more bytes at offsets 36 and 20 may overlap 1 byte at offset 36 [-Wrestrict]
  strcat(card->shortname, card->driver);

It happens since gcc calculates the possible string size from
card->driver, but this can't be true since we did set the string just
before that, and they are much shorter.

For shutting it up, use the exactly same string set to card->driver
for strcat() to card->shortname, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 15:01:37 +02:00
Takashi Iwai 63623646a0 ALSA: emu10k1: Fix missing __force annotation for user/kernel pointer cast
The cast between user-space and kernel-space needs an explicit __force
prefix, but it's missing in many places in emu10k1 driver code.

Spotted by sparse as a warning like:
  sound/pci/emu10k1/emufx.c:529:33: warning: cast removes address space of expression

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:33:16 +02:00
Takashi Iwai 0701492c86 ALSA: korg1212: Add __force annotation to cast in user-copy callbacks
The user-copy callbacks in korg1212 driver contain the explicit cast
from a user pointer to a kernel pointer, but they missed __force
prefix.  It's mandatory for converting between them.

Spotted by sparse, a warning like:
  sound/pci/korg1212/korg1212.c:1329:33: warning: cast removes address space of expression

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:33:08 +02:00
Takashi Iwai 191bb51e72 ALSA: pcm: Use standard lower_32_bits() and upper_32_bits()
Instead of open codes, use the standard macros for obtaining the lower
and upper 32bit values.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:32:31 +02:00
Takashi Iwai 00966dcdf0 ALSA: usb-audio: Declare the common variable in header file
Declare snd_usb_feature_unit_ctl properly in mixer.h.  Otherwise it's
error-prone.

This fixes the sparse warning:
  sound/usb/mixer.c:1464:25: warning: symbol 'snd_usb_feature_unit_ctl' was not declared. Should it be static?

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:32:00 +02:00
Takashi Iwai 7e9c20f403 ALSA: opl3: Declare common variables properly
Move the declarations of common variables into opl3_voice.h instead of
declaring at each file multiple times, which was error-prone.

This fixes sparse warnings like:
  sound/drivers/opl3/opl3_synth.c:51:6: warning: symbol 'snd_opl3_regmap' was not declared. Should it be static?

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:31:48 +02:00
Takashi Iwai ebd836edfc ALSA: hda - Fix a sparse warning about snd_ctl_elem_iface_t
The knew->iface field is in snd_ctl_elem_iface_t, which is with
__bitwise, hence it can't be converted implicitly from integer.
Give an explicit cast for the invalid type.

Spotted by sparse:
  sound/pci/hda/hda_codec.c:3280:25: warning: restricted snd_ctl_elem_iface_t degrades to integer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:31:41 +02:00
Takashi Iwai dcda6f7853 ALSA: msnd: Use NULL instead of 0
Fix a sparse warning:
  sound/isa/msnd/msnd_pinnacle.c:813:1: warning: Using plain integer as NULL pointer

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:31:33 +02:00
Takashi Iwai bb86124c80 ALSA: hda/ca0132 - Use NULL instead of 0
Use NULL for initializing the snd_kcontrol_new.tlv field, instead of
0, as warned by sparse:
  sound/pci/hda/patch_ca0132.c:5519:22: warning: Using plain integer as NULL pointer

Also, the driver does the same initialization twice, once for
knew.tlv.c and another for knew.tlv.p while both point to the same
address (these are union).  Drop the latter superfluous one.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:31:25 +02:00
Takashi Iwai 7c500f9ea1 ALSA: msnd: Fix the default sample sizes
The default sample sizes set by msnd driver are bogus; it sets ALSA
PCM format, not the actual bit width.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:31:15 +02:00
Takashi Iwai ab647a2d62 ALSA: msnd: Add missing __iomem annotations
The io-mapped buffers used in msnd drivers need __iomem annotations.

This fixes sparse warnings like:
  sound/isa/msnd/msnd_pinnacle.c:172:45: warning: incorrect type in initializer (different address spaces)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:31:06 +02:00
Takashi Iwai bd1cd0eb2c ALSA: usb-audio: Fix multiple definitions in AU0828_DEVICE() macro
AU0828_DEVICE() macro in quirks-table.h uses USB_DEVICE_VENDOR_SPEC()
for expanding idVendor and idProduct fields.  However, the latter
macro adds also match_flags and bInterfaceClass, which are different
from the values AU0828_DEVICE() macro sets after that.

For fixing them, just expand idVendor and idProduct fields manually in
AU0828_DEVICE().

This fixes sparse warnings like:
  sound/usb/quirks-table.h:2892:1: warning: Initializer entry defined twice

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:30:57 +02:00
Jeff Crukley b080dc5bd0 ALSA: usb-audio: Add support for Encore mDSD USB DAC
This patch adds native DSD playback support for the Encore mDSD USB DAC by
specifying the vendor and product ID's

Signed-off-by: Jeff Crukley <jcrukley@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:29:32 +02:00
Takashi Iwai 69756930f2 ALSA: cs5535audio: Fix invalid endian conversion
One place in cs5535audio_build_dma_packets() does an extra conversion
via cpu_to_le32(); namely jmpprd_addr is passed to setup_prd() ops,
which writes the value via cs_writel().  That is, the callback does
the conversion by itself, and we don't need to convert beforehand.

This patch fixes that bogus conversion.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:23:45 +02:00
Takashi Iwai 3acd3e3bab ALSA: vxpocket: Fix invalid endian conversions
The endian conversions used in vxp_dma_read() and vxp_dma_write() are
superfluous and even wrong on big-endian machines, as inw() and outw()
already do conversions.  Kill them.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:23:33 +02:00
Takashi Iwai fff71a4c05 ALSA: vx222: Fix invalid endian conversions
The endian conversions used in vx2_dma_read() and vx2_dma_write() are
superfluous and even wrong on big-endian machines, as inl() and outl()
already do conversions.  Kill them.

Spotted by sparse, a warning like:
  sound/pci/vx222/vx222_ops.c:278:30: warning: incorrect type in argument 1 (different base types)

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:23:28 +02:00
Takashi Iwai a49a71f6e2 ALSA: seq: Fix poll() error return
The sanity checks in ALSA sequencer and OSS sequencer emulation codes
return falsely -ENXIO from poll callback.  They should be EPOLLERR
instead.

This was caught thanks to the recent change to the return value.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-26 08:23:26 +02:00
Rakesh Ughreja fe65324e3f
ASoC: Intel: Skylake: fix widget handling
include DAPM Mux and output widgets into the list.

Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja@intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-25 17:30:30 +01:00
Pierre-Louis Bossart 9a0daaab31
ASoC: Intel: Atom: fix inversion between __iowrite32 and __ioread32
This looks like a copy/paste issue, but clearly there is an inversion
that is obvious when checking the arguments.

Detected with Sparse - now that we have fewer warnings this one was
easy to find.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-25 17:21:22 +01:00
Pierre-Louis Bossart ce1cfe295a
ASoC: Intel: Atom: simplify iomem address and casts
Simplify code and add relevant casts to make Sparse warnings go away

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-25 17:21:18 +01:00
Pierre-Louis Bossart ef3cb74233
ASoC: Intel: common: make sst_dma functions static
sst_dma_new and sst_dma_free are not used in any other file and don't
have a prototype. Move to static functions and remove
EXPORT_SYMBOL_GPL statement.

Reported by sparse warnings.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-25 17:21:15 +01:00
Pierre-Louis Bossart 86efd35ec1
ASoC: Intel: Skylake: BDL definitions should be __le32
Make sure definitions are consistent with usage.
Detected with Sparse.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-25 17:21:11 +01:00
Pierre-Louis Bossart 92beb0a269
ASoC: Intel: Haswell: fix endianness handling
Make all Sparse warnings go away by using le16/32_to_cpu.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-25 17:21:07 +01:00
Bard Liao d77a760842
ASoC: rt5631: add Volume to the name of volume control
add Volume to the name of volume control.

Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-25 17:13:40 +01:00
Fabio Estevam 6b24e03ecf
ASoC: imx-sgtl5000: Switch to SPDX identifier
Adopt the SPDX license identifier headers to ease license compliance
management.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-24 16:59:07 +01:00
Fabio Estevam ad47191a72
ASoC: fsl_utils: Switch to SPDX identifier
Adopt the SPDX license identifier headers to ease license compliance
management.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-24 16:59:03 +01:00
Fabio Estevam 2ba2805368
ASoC: fsl_asrc: Switch to SPDX identifier
Adopt the SPDX license identifier headers to ease license compliance
management.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-24 16:58:59 +01:00
Fabio Estevam aa624a0a92
ASoC: fsl-asoc-card: Switch to SPDX identifier
Adopt the SPDX license identifier headers to ease license compliance
management.

Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-24 16:58:55 +01:00
Srinivas Kandagatla 467b061f1a
ASoC: core: add support to snd_soc_dai_get_channel_map()
On Qualcomm platforms, specifically with SLIMbus interfaced codecs,
the codec slim channel numbers are passed to DSP while configuring
the slim audio path. Having get_channel_map() would allow dais to
share such information across multiple dais.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-24 12:06:43 +01:00
Oder Chiou d96f8bd28c
ASoC: rt5514: Fix the issue of the delay volume applied
The patch fixes the issue of the delay volume applied.

Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-24 11:57:05 +01:00
Arnd Bergmann a241c3d95b
ASoC: meson: axg-spdifout: select SND_PCM_IEC958
When CONFIG_SND_PCM_IEC958 is disabled, we get a link error for the
new driver:

sound/soc/meson/axg-spdifout.o: In function `axg_spdifout_hw_params':
axg-spdifout.c:(.text+0x650): undefined reference to `snd_pcm_create_iec958_consumer_hw_params'

The other users use 'select', so we should do the same here.

Fixes: 53eb4b7aaa ("ASoC: meson: add axg spdif output")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-24 11:56:28 +01:00
Takashi Iwai f9b54e1961 ALSA: hda/i915: Allow delayed i915 audio component binding
Currently HD-audio i915 audio binding doesn't support any delayed
binding, and supposes that the i915 driver registers the component
immediately.  This has been OK, so far, but the work-in-progress
change in i915 may introduce the asynchronous binding, which
effectively delays the component registration.

For addressing it, implement a completion to be synced with the master
binding.  The timeout is set to 10 seconds which should be long enough
and hopefully be not too annoying if anyone boots up a debugging
session with i915 KMS turned off.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-24 12:32:44 +02:00
Yue Wang 1ea0358ecb ALSA: usb-audio: Generic DSD detection for Thesycon-based implementations
Thesycon provides solutions to XMOS chips, and has its own device
vendor id.

In this patch, we use generic method to detect DSD capability of
Thesycon-based UAC2 implementations in order to support a wide range
of current and future devices.

The patch will enable the SNDRV_PCM_FMTBIT_DSD_U32_BE bit for the DAC
hence enable native DSD playback up to DSD512 format.

Signed-off-by: Yue Wang <yuleopen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-23 11:29:59 +02:00
Takashi Iwai dfef01e150 ALSA: memalloc: Don't exceed over the requested size
snd_dma_alloc_pages_fallback() tries to allocate pages again when the
allocation fails with reduced size.  But the first try actually
*increases* the size to power-of-two, which may give back a larger
chunk than the requested size.  This confuses the callers, e.g. sgbuf
assumes that the size is equal or less, and it may result in a bad
loop due to the underflow and eventually lead to Oops.

The code of this function seems incorrectly assuming the usage of
get_order().  We need to decrease at first, then align to
power-of-two.

Reported-and-tested-by: he, bo <bo.he@intel.com>
Reported-by: zhang jun <jun.zhang@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-23 09:06:33 +02:00
Srikanth K H d10ee9c542 ALSA: timer: catch invalid timer object creation
A timer object for the classes SNDRV_TIMER_CLASS_CARD and
SNDRV_TIMER_CLASS_PCM has to be associated with a card object, but we
have no check at creation time.  Such a timer object with NULL card
causes various unexpected problems, e.g. NULL dereference at reading
the sound timer proc file.

So as preventive measure while the creating the sound timer object is
created the card information availability is checked for the mentioned
entries and returned error if its NULL.

Signed-off-by: Srikanth K H <srikanth.h@samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-22 10:42:41 +02:00
Daniel Mack a8e43c21a8
ASoC: pxa: remove clock divider and pll setup from zylonite and magician
The SSP DAI now handles the clocking setup itself, all it needs is the
master clock frequency. Remove the code from Zylonite and Magician
platforms.

Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 17:41:26 +01:00
Jerome Brunet 7864a79f37
ASoC: meson: add axg sound card support
Add the axg sound card to handle the specifities of the axg audio
sub system.

This card is required to:
 * setup the dpcm links specific to the AXG (with a cpu sound dai)
 * handle the 4 lanes masks of the tdm interfaces
 * add the loopback link when a tdm pad interface has a playback
   stream
 * handle multi-codec links

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 17:40:12 +01:00
Jerome Brunet cbdfab3b67
ASoC: export snd_soc_of_get_slot_mask
Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot()
directly because it needs to handle 4 mask for each direction.
Yet the parsing of each mask is the same, so export
snd_soc_of_get_slot_mask() to reuse the the existing code.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 17:39:26 +01:00
Jerome Brunet 13a22e6a98
ASoC: meson: add tdm input driver
Add Amlogic's axg TDM input driver which take the TDM signal of 4 input
lanes and push the decoded audio samples to TODDR fifo

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 17:39:04 +01:00
Jerome Brunet c41c2a355b
ASoC: meson: add tdm output driver
Add Amlogic's axg tdm output driver which pulls data from FRDDR fifo
and produce the TDM signals for 4 output lanes.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 17:38:55 +01:00
Jerome Brunet d60e4f1e4b
ASoC: meson: add tdm interface driver
Add Amlogic's axg TDM interface driver. This driver manages the format
and clocks provided on the pads.

On this SoC, each stream direction provides 4 serial lanes. This makes
a maximum of 8 channels in i2s modes and 128 channels in DSP modes.

While each lanes operate on the same slot number (same bit clock), they
may have different TDM masks. This requires to provide a function to let
the card set the 4 masks, in lieu of the usual set_tdm_slots() callback
of the dai driver.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 17:38:33 +01:00
Jerome Brunet 1a11d88f49
ASoC: meson: add tdm formatter base driver
Add Amlogic's axg TDM core driver. On this SoC, tdm is bit more
complex than usual, mainly because the different TDM input decoders can
be attached to any of TDM pad interface, including the output pads.

For the this, TDM on this SoC is modeled like this:
- TDM interface provides the DAIs the codecs will be attached to.
  The main responsibility of this driver is to manage the pad format
  and the TDM clock rates.
- TDM Formatters: These are the entities which are actually dealing with
  the TDM signal. TDMOUT produce a TDM signal from the audio sample
  provided by FRDDR using the clocks provided the TDM interface. TDMIN
  feeds TODDR with audio sample using the clocks and TDM signal provided
  by the TDM Interface.
- TDM Streams: This provides the link between 1 DAI stream of the TDM
  interface and one (or more) TDM formatters.

This driver provides the TDM formatter and TDM stream operations.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 17:38:27 +01:00
Marcel Ziswiler 2ec4248635
ASoC: tegra: improve goto error label
While the two error labels "err" and "err_clk_put" goto the same place
it is rather confusing that the earlier one is certainly used later
again.

Signed-off-by: Marcel Ziswiler <marcel.ziswiler@toradex.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-20 13:18:32 +01:00
Geoff Levand 48e9184686 powerpc/ps3: Set driver coherent_dma_mask
Set the coherent_dma_mask for the PS3 ehci, ohci, and snd devices.

Silences WARN_ON_ONCE messages emitted by the dma_alloc_attrs() routine.

Reported-by: Fredrik Noring <noring@nocrew.org>
Signed-off-by: Geoff Levand <geoff@infradead.org>
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Acked-by: Alan Stern <stern@rowland.harvard.edu>
Signed-off-by: Michael Ellerman <mpe@ellerman.id.au>
2018-07-20 12:50:37 +10:00
Takashi Iwai 7abeb64da6 Merge branch 'topic/drm_audio_component' into for-next
Pull the generic drm_audio_component support, which will be used later
for AMD/ATI and other HD-audio HDMI codec drivers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-19 20:48:14 +02:00
Hans de Goede 9ee6f8a8cb
ASoC: Intel: bytcr_rt5640: Add quirk for the "Connect Tablet 9" tablet
Add a quirk for the "Connect Tablet 9" tablet, this tablet has a
mono-speaker. Otherwise it works fine with the defaults.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 18:08:26 +01:00
Hans de Goede 06aa6e5127
ASoC: Intel: bytcr_rt5651: Add quirk table entries for various devices
Add quirk table entries for the following tablets:

ITWorks TW701
Ployer Momo7w
Trekstor win7
Yours 8"

These all use the default settings, except that they only have a single
speaker and thus need the mono-speaker quirk.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 16:04:41 +01:00
Hans de Goede a0d1d867c2
ASoC: Intel: bytcr_rt5651: Add mono speaker quirk
During my initial round of bytcr_rt5651 long-name patches I did not include
a difference for mono vs stereo speaker setups in the longname because it
seems that all 5651 devices with only a single speaker do some mixing of
left + right on the PCB.

However further testing has shown that while this works great when only
playing audio on the left or right channel, the output becomes garbled
when using both channels at once. Something which does not happen when
using the Stereo DAC MIXL / MIXR switches to mix the channels together
inside the codec and then only outputting on a single channel.

So we need to have separate UCM profiles and thus separate long-names
for devices with a mono speaker vs stereo speakers. Just as we already
have for the bytcr_rt5640 case.

This commit adds a new BYT_RT5651_MONO_SPEAKER quirk and adds "stereo-spk"
or "mono-spk" to the long-name based on this and enables this mapping on
devices with a mono speaker.

Changing the long-name like this is ok for now, since I'm still working
on the UCM profiles, so they are not in upstream alsa-lib yet.

This brings the long-name naming scheme fully in sync with the bytcr_rt5640
case, which is good from a consistency pov.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 16:04:35 +01:00
Hans de Goede ac275ee5aa
ASoC: Intel: bytcr_rt5651: Add IN2 input mapping
During the recent cleanup series 3 of the 6 input mappings where removed
from the bytcr_rt5651 machine driver because testing showed that none of
them were used.

However some devices do actually have their internal mic on IN2 (and
only IN2, not IN1 and IN2), this did not show during previous tests
due to a bug in the userspace UCM input device switching code.

This commit re-adds the IN2 mapping for devices with the internal mic.
on IN2 and the headser mic on IN3 and enables this mapping on devices
with their internal mic on IN2.

This commit also changes the default internal mic input to IN2, because
all my 7 test devices have their mic there.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 16:04:32 +01:00
Hans de Goede 8627fb257e
ASoC: Intel: bytcr_rt5651: Set OVCD limit for VIOS LTH17 to 2000uA
With the default over current detect limit of 1500uA headsets on often
get detected as headphones on the VIOS LTH17 and even when detected as
headset the OVCD current triggers often while plugged in, resulting in
false-positive button press detection.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 16:04:28 +01:00
Hans de Goede 0a3badd141
ASoC: Intel: bytcr_rt5651: Fix using the wrong GPIO for the ext-amp on some boards
Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
boards may  have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
GpioIo one for the ext-amp-enable-gpio.

So far we've been assuming that the GpioIo one always comes first, this
commit adds code to detect which one comes first and to add the right
gpio-mapping.

This fixes sound not working on the Vios LTH17 laptop.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 16:04:13 +01:00
Hans de Goede eea1662525
ASoC: rt5651: Add IN3 Boost volume control
Add a mixer control for the IN3 Boost volume, IN3 is used for the headset
mic on most devices, so this is necessary to control the headset mic
volume.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 16:02:55 +01:00
Richard Fitzgerald d52ed4b0bc
ASoC: wm_adsp: Parse HOST_BUFFER controls
Currently the compressed streams in DSP firmwares are
identified essentially by looking at a fixed location inside
the firmware. This is fragile and also limits things to a
single compressed stream.

Here a new form of firmware parameter is added, the HOST_BUFFER
which identifies a compressed stream from meta-data in the
firmware file. This is more robust and allows for the possiblity
of using multiple streams per core in the future. Currently the
implementation is still limited to a single stream and will
use the first HOST_BUFFER parameter encountered. If there aren't
any HOST_BUFFER parameters it will fall back to the legacy way
of finding the host buffer.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 15:11:56 +01:00
Richard Fitzgerald 3bbc2705a3
ASoC: wm_adsp: Allow up to 8 channels for voice control
Newer voice control firmwares can capture multiple audio channels.
Allow up to 8 channels for future-proofing.

Signed-off-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 15:11:55 +01:00
Charles Keepax b7ede5af62
ASoC: wm_adsp: Take prefix into account in control name length
Currently when creating ALSA control names for the DSP the length of any
prefix applied to the CODEC is not taken into account. Whilst this is
mostly harmless it does result in ALSA doing the truncation of the
control names and printing a warning. It is better to have the driver do
the truncation so it can truncate from the start of parameter name
itself to give a greater chance of the result maintain a unique name.

Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 15:11:54 +01:00
Charles Keepax 517ee74e1b
ASoC: wm_adsp: Correct algorithm list allocation size
Commit 6396bb2215 ("treewide: kzalloc() -> kcalloc()") was
overlooked when doing some refactoring to the algorithm list
handling, which lead to twice as much buffer being allocated
as required for reading the algorithm list. A kcalloc is no
longer appropriate since the allocation size is now in bytes
not registers, as such change back to kzalloc.

Fixes: 7f7cca08ab ("ASoC: wm_adsp: Simplify handling of alg offset and length")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 15:11:53 +01:00
Stuart Henderson 868e49a4a0
ASoC: wm_adsp: Ensure DSP boot work complete before preloader_put return
All controls derived from the loaded firmware should be created prior
to returning from the preloader's put function, such that they are
immediately available to user-space.

Signed-off-by: Stuart Henderson <stuarth@opensource.cirrus.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-19 15:09:41 +01:00
Adam Goode 58cabe8715 ALSA: usb-audio: Allow changing from a bad sample rate
If the audio device is externally clocked and set to a rate that does
not match the external clock, the clock will never be valid and we cannot
set the rate successfully. To fix this, allow a rate change even if
the clock is initially invalid, and validate again after the rate is
changed.

This fixes problems with MOTU UltraLite AVB hardware over USB.

Signed-off-by: Adam Goode <agoode@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-19 08:44:46 +02:00
Takashi Iwai 67ece13ffe Merge branch 'topic/vga_switcheroo' into for-next
Pull the vga_switcheroo audio client fix.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-18 17:42:40 +02:00
Jerome Brunet 53eb4b7aaa
ASoC: meson: add axg spdif output
Add support for the spdif output serializer of the axg SoC family

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:39 +01:00
Jerome Brunet 7ed4877b40
ASoC: meson: add axg toddr driver
Add the capture memory interface of Amlogic's axg SoCs.
TDM, SPDIF or PDM input devices place audio samples inside this FIFO.
The FIFO content is then pushed to DDR

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:37 +01:00
Jerome Brunet 57d552e3ea
ASoC: meson: add axg frddr driver
Add the playback memory interface of Amlogic's axg SoCs.
This device pulls data from DDR to an internal FIFO.
This FIFO is then used to feed TDM and SPDIF Output devices.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:37 +01:00
Jerome Brunet 6dc4fa179f
ASoC: meson: add axg fifo base driver
Amlogic's axg SoCs have two types of fifos which are the memory
interfaces of the audio subsystem. FRDDR provides the playback
interface while TODDR provides the capture interface.

The way these fifos operate is very similar. Only a few settings
are specific to each.

They implement the same pcm driver here and the specifics of each
will be dealt with the related DAI driver.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:36 +01:00
Sriram Periyasamy bf270262b7
ASoC: hdac_hdmi: Add documentation for power management
Add documentation for power management of HDAC HDMI codec device for
various scenarios such as S0/S3, probe and playback use case.

Signed-off-by: Sriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: Sanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:34 +01:00
Srinivas Kandagatla 90ae7105ea
ASoC: qcom: apq8096: remove component framework related code
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:30 +01:00
Srinivas Kandagatla 791940779d
ASoC: qdsp6: q6routing: remove component framework related code
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:26 +01:00
Srinivas Kandagatla f924e4fd89
ASoC: qdsp6: q6asm-dai: remove component framework related code
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:23 +01:00
Srinivas Kandagatla 605fcb6991
ASoC: qdsp6: q6afe-dai: remove component fw related code
Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:19 +01:00
Srinivas Kandagatla bb4b894add
ASoC: core: add support to card re-bind using component framework
This patch aims at achieving dynamic behaviour of audio card when
the dependent components disappear and reappear.

With this patch the card is removed if any of the dependent component
is removed and card is added back if the dependent component comes back.
All this is done using component framework and matching based on
component name.

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-18 13:08:15 +01:00
Takashi Iwai f3d737b634 ALSA: hda/realtek - Yet another Clevo P950 quirk entry
The PCI SSID 1558:95e1 needs the same quirk for other Clevo P950
models, too.  Otherwise no sound comes out of speakers.

Bugzilla: https://bugzilla.opensuse.org/show_bug.cgi?id=1101143
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-18 12:17:46 +02:00
Takashi Iwai fa84cf094e ALSA: pcm: Nuke snd_pcm_lib_mmap_vmalloc()
snd_pcm_lib_mmap_vmalloc() was supposed to be implemented with
somewhat special for vmalloc handling, but in the end, this turned to
just the default handler, i.e. NULL.  As the situation has never
changed over decades, let's rip it off.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-18 08:24:29 +02:00
Takashi Iwai ef4db239cd ALSA: rawmidi: Use kvmalloc() for buffers
The size of in-kernel rawmidi buffers may be big up to 1MB, and it can
be specified freely by user-space; which implies that user-space may
trigger kmalloc() errors frequently.

This patch replaces the buffer allocation via kvmalloc() for dealing
with bigger buffers gracefully.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-18 07:47:57 +02:00
Takashi Iwai f5beb598b0 ALSA: rawmidi: Minor code refactoring
Unify a few open codes with helper functions to improve the
readability.  Minor behavior changes (rather fixes) are:
- runtime->drain clearance is done within lock
- active_sensing is updated before resizing buffer in
  SNDRV_RAWMIDI_IOCTL_PARAMS ioctl.
Other than that, simply code cleanups.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 23:07:29 +02:00
Takashi Iwai 7fdc9b0807 ALSA: rawmidi: Simplify error paths
Apply the standard idiom: rewrite the multiple unlocks in error paths
in the goto-error-and-single-unlock way.

Just a code refactoring, and no functional changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 22:48:38 +02:00
Takashi Iwai 5bed913972 ALSA: rawmidi: Tidy up coding styles
Just minor coding style fixes like removal of superfluous white space,
adding missing blank lines, etc.  No actual code changes at all.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 22:37:07 +02:00
Takashi Iwai ed6b83d2d1 Merge branch 'for-linus' into for-next
Back-merge for further cleanup / improvements on rawmidi and HD-audio
stuff.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 22:27:03 +02:00
Takashi Iwai a57942bfdd ALSA: hda: Make audio component support more generic
This is the final step for more generic support of DRM audio
component.  The generic audio component code is now moved to its own
file, and the symbols are renamed from snd_hac_i915_* to
snd_hdac_acomp_*, respectively.  The generic code is enabled via the
new kconfig, CONFIG_SND_HDA_COMPONENT, while CONFIG_SND_HDA_I915 is
kept as the super-class.

Along with the split, three new callbacks are added to audio_ops:
pin2port is for providing the conversion between the pin number and
the widget id, and master_bind/master_unbin are called at binding /
unbinding the master component, respectively.  All these are optional,
but used in i915 implementation and also other later implementations.

A note about the new snd_hdac_acomp_init() function: there is a slight
difference between this and the old snd_hdac_i915_init().  The latter
(still) synchronizes with the master component binding, i.e. it
assures that the relevant DRM component gets bound when it returns, or
gives a negative error.  Meanwhile the new function doesn't
synchronize but just leaves as is.  It's the responsibility by the
caller's side to synchronize, or the caller may accept the
asynchronous binding on the fly.

v1->v2: Fix missing NULL check in master_bind/unbind

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 22:25:48 +02:00
Takashi Iwai 82887c0beb ALSA: hda/i915: Associate audio component with devres
The HD-audio i915 binding code contains a single pointer, hdac_acomp,
for allowing the access to audio component from the master bind/unbind
callbacks.  This was needed because the callbacks pass only the device
pointer and we can't guarantee the object type assigned to the drvdata
(which is free for each controller driver implementation).
And this implementation will be a problem if we support multiple
components for different DRM drivers, not only i915.

As a solution, allocate the audio component object via devres and
associate it with the given device, so that the component callbacks
can refer to it via devres_find().

The removal of the object is still done half-manually via
devres_destroy() to make the code consistent (although it may work
without the explicit call).

Also, the snd_hda_i915_register_notifier() had the reference to
hdac_acomp as well.  In this patch, the corresponding code is removed
by passing hdac_bus object to the function, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 22:25:47 +02:00
Takashi Iwai ae891abe7c drm/i915: Split audio component to a generic type
For allowing other drivers to use the DRM audio component, rename the
i915_audio_component_* with drm_audio_component_*, and split the
generic part into drm_audio_component.h.  The i915 specific stuff
remains in struct i915_audio_component, which contains
drm_audio_component as the base.

The license of drm_audio_component.h is kept to MIT as same as the the
original i915_component.h.

This is a preliminary change for further development, and no
functional changes by this patch itself, merely code-split and
renames.

v1->v2: Use SPDX for drm_audio_component.h, fix remaining i915
        argument in drm_audio_component.h

Reviewed-by: Rodrigo Vivi <rodrigo.vivi@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 22:25:19 +02:00
Takashi Iwai 39675f7a7c ALSA: rawmidi: Change resized buffers atomically
The SNDRV_RAWMIDI_IOCTL_PARAMS ioctl may resize the buffers and the
current code is racy.  For example, the sequencer client may write to
buffer while it being resized.

As a simple workaround, let's switch to the resized buffer inside the
stream runtime lock.

Reported-by: syzbot+52f83f0ea8df16932f7f@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 17:33:17 +02:00
Gustavo A. R. Silva 7373c2a99a ALSA: emu8000: Use swap macro in snd_emu8000_sample_new
Make use of the swap macro and remove unnecessary variable *tmp*. This
makes the code easier to read and maintain.

This code was detected with the help of Coccinelle.

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 17:18:03 +02:00
Gustavo A. R. Silva e2d2f24049 ALSA: emu10k1_patch: Use swap macro in snd_emu10k1_sample_new
Make use of the swap macro and remove unnecessary variable *tmp*. This
makes the code easier to read and maintain.

This code was detected with the help of Coccinelle.

Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 17:17:52 +02:00
Jim Qu 4aaf448fa9 vga_switcheroo: set audio client id according to bound GPU id
On modern laptop, there are more and more platforms
have two GPUs, and each of them maybe have audio codec
for HDMP/DP output. For some dGPU which is no output,
audio codec usually is disabled.

In currect HDA audio driver, it will set all codec as
VGA_SWITCHEROO_DIS, the audio which is binded to UMA
will be suspended if user use debugfs to contorl power

In HDA driver side, it is difficult to know which GPU
the audio has binded to. So set the bound gpu pci dev
to vga_switcheroo.

if the audio client is not the third registration, audio
id will set in vga_switcheroo enable function. if the
audio client is the last registration when vga_switcheroo
_ready() get true, we should get audio client id from bound
GPU directly.

Signed-off-by: Jim Qu <Jim.Qu@amd.com>
Reviewed-by: Lukas Wunner <lukas@wunner.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-17 11:12:00 +02:00
YOKOTA Hiroshi 0fca97a29b ALSA: hda/realtek - Add Panasonic CF-SZ6 headset jack quirk
This adds some required quirk when uses headset or headphone on
Panasonic CF-SZ6.

Signed-off-by: YOKOTA Hiroshi <yokota.hgml@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:48:50 +02:00
Po-Hsu Lin 9a6249d2a1 ALSA: hda: add mute led support for HP ProBook 455 G5
Audio mute led does not work on HP ProBook 455 G5,
this can be fixed by using CXT_FIXUP_MUTE_LED_GPIO to support it.

BugLink: https://bugs.launchpad.net/bugs/1781763
Reported-by: James Buren
Signed-off-by: Po-Hsu Lin <po-hsu.lin@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:47:42 +02:00
Jim Qu b6d7b3622b ALSA: hda: use PCI_BASE_CLASS_DISPLAY to replace PCI_CLASS_DISPLAY_VGA
Except PCI_CLASS_DISPLAY_VGA, some PCI class is sometimes
PCI_CLASS_DISPLAY_3D or PCI_CLASS_DISPLAY_OTHER.

Signed-off-by: Jim Qu <Jim.Qu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:37:08 +02:00
Jorge Sanjuan 55b8cb46a7 ALSA: usb-audio: Tidy up logic for Processing Unit min/max values
This patch refactors the processing units min/max calculation logic
for the mixer controls and fixes an issue where the Mode Select
checking of the Up/Down mixers doesn't differentiate between the
UAC1 and UAC2 Control Selector (0x02) and the UAC3 one which is
different (0x01).

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:36:15 +02:00
Jorge Sanjuan 8b3a087f7f ALSA: usb-audio: Unify virtual type units type to UAC3 values
The Audio Control interface descriptor subtypes do not match
across all the UAC versions. That makes reusability of the
"virtual type" (Mixer, Processors, Selectors, etc) terminals
difficult. It also makes the mixer get the default names for
the virtual terminals wrong due to the overlap.

This patch proposes an unified approach by always using the most
comprehensive spec version to define them all (in this case UAC3).

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:35:55 +02:00
Jorge Sanjuan 0f292f023f ALSA: usb-audio: Add support for Processing Units in UAC3
This patch adds support for the Processig Units defined in
the UAC3 spec. The main difference with the previous specs
is the lack of on/off switches in the controls for these
units and the addiction of the new Multi Function Processing
Unit.

The current version of the UAC3 spec doesn't define any
useful controls for the new Multi Function Processing Unit
so no control will get created once this unit is parsed.

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:35:34 +02:00
Jorge Sanjuan 4e887af31c ALSA: usb-audio: Processing Unit controls parsing in UAC2
Current support for UAC2 Processing Units does the parsing
as one control per bit in the bitmap. However, the UAC2 spec
defines the controls as bit pairs where b01 means read-only
and b11 means read/write control.

This patch fixes that and uses the helper functions for checking
controls readability/writability when the control is defined as
bit pairs (UAC2 and UAC3).

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:35:09 +02:00
Jorge Sanjuan c77e1ef1cd ALSA: usb-audio: Add support for Selector Units in UAC3
This patch add support for Selector Units and Clock Selector Units
defined in the new UAC3 spec.

Selector Units play a really important role in the new UAC3 spec as
Processing Units do not define an on/off switch control anymore.
This forces topology designers to add bypass paths in the topology
to enable/dissable the Processing Units.

Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 16:34:44 +02:00
Agrawal, Akshu 19e023e3be
ASoC: AMD: For capture have interrupts on I2S->ACP channel
Having interrupts enabled for ACP<->SYSMEM DMA transfer, we are in
for an interrupt storm.
For both playback and capture interrupts should be enabled for
I2S<->ACP DMA.

Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-16 15:30:12 +01:00
Agrawal, Akshu fa9d2f17c2
ASoC: AMD: Send correct channel for configuring DMA descriptors
Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for
ACP-I2S transfer. With recent patches ch1 is used to define channel
order number 1 and ch2 as channel order number 2. Thus,
Playback:
ch1:SYSMEM->ACP
ch2:ACP->I2S
Capture:
ch1:I2S->ACP
ch1:ACP->SYSMEM

Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-16 15:30:11 +01:00
Naveen Manohar 8452112baa
ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver
Patch adds Geminilake I2S machine driver which uses following codecs:
RT5682 and MAX98357A.

Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-16 15:30:10 +01:00
Russell King d30e23d699
ASoC: hdmi-codec: fix routing
Commit 943fa02282 ("ASoC: hdmi-codec: Use different name for playback
streams") broke hdmi-codec's routing between it's output "TX" widget
and the S/PDIF or I2S streams by renaming the streams.

Whether an error occurs or not is dependent on whether there is another
widget called "Playback" registered by some other component - if there
is, that widget will be (incorrectly) bound to the HDMI codec's "TX"
output widget.  If we end up connecting "TX" incorrectly, it can result
in components not being started, causing no audio output.

Since the I2S and S/PDIF streams now have different names, we can't
use a static route at component level to describe the relationship, so
arrange to dynamically create the route when the DAI driver is probed.

Fixes: 943fa02282 ("ASoC: hdmi-codec: Use different name for playback streams")
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-16 15:30:03 +01:00
Colin Ian King d6e08c7eab ALSA: cs46xx: remove redundant pointer 'ins'
Pointer 'ins' is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'ins' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 14:30:46 +02:00
Colin Ian King c888443951 ALSA: ali5451: remove redundant pointer 'codec'
Pointer 'codec' is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'codec' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 14:30:33 +02:00
Colin Ian King 7527cd209e ALSA: sb8: remove redundant pointer runtime
Pointer runtime is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'runtime' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 14:30:20 +02:00
Colin Ian King 29fba9230d ALSA: gus: remove redundant pointer private_data
Pointer private_data is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'private_data' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 14:29:57 +02:00
Colin Ian King a34e8aac49 ALSA: es1688: remove redundant pointer chip
Pointer chip is being assigned but is never used hence it is
redundant and can be removed.

Cleans up clang warning:
warning: variable 'chip' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 14:29:46 +02:00
Colin Ian King eeef847de5 ALSA: opl3: remove redundant pointer opl3
Variable opl3 is being assigned but is never used hence it is
redundant and can be removed.

Cleans up several clang warnings:
warning: variable 'opl3' set but not used [-Wunused-but-set-variable]

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-16 14:29:37 +02:00
Dan Carpenter 090345ce72
ASoC: qdsp6: q6routing: off by one in routing_hw_params()
The data->port_data[] array has AFE_MAX_PORTS elements so the check
should be >= instead of > or we write one element beyond the end of the
array.

Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-13 16:26:20 +01:00
Dan Carpenter b8110a87b7
ASoC: qdsp6: q6afe-dai: fix a range check in of_q6afe_parse_dai_data()
The main thing is that the data->priv[] array has AFE_PORT_MAX elements
so the > condition should be >=.  But we may as well check for negative
values as well just to be safe.

Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-13 16:26:09 +01:00
Jerome Brunet aefba45539
ASoC: allow soc-core to pick up name prefixes from component nodes
When the component does not match the configuration table provided
by the card, let soc-core check the component node for a name prefix

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-13 16:05:28 +01:00
Takashi Iwai 9a9b13dd27 Merge branch 'topic/hda-core-intel' into topic/hda-acomp 2018-07-12 13:58:07 +02:00
Alastair Bridgewater c5a59d2477 ALSA: hda/ca0132: Update a pci quirk device name
The PCI subsystem in question for this quirk rule has been
identified as a Gigabyte GA-Z170X-Gaming 7 motherboard.  Set the
device name appropriately.

Signed-off-by: Alastair Bridgewater <alastair.bridgewater@gmail.com>
Reviewed-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-12 09:18:31 +02:00
Alastair Bridgewater dad59262b7 ALSA: hda/ca0132: Add Recon3Di quirk for Gigabyte G1.Sniper Z97
These motherboards have Sound Core3D and apparently "support"
Recon3Di.  Added to the quirk list as QUIRK_R3DI.

Issue report, PCI Subsystem ID, and testing by a contributor on
IRC who wished to remain anonymous.

Signed-off-by: Alastair Bridgewater <alastair.bridgewater@gmail.com>
Reviewed-by: Connor McAdams <conmanx360@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-12 09:18:08 +02:00
Jerome Brunet baacd8d100
ASoC: dpcm: add rate merge to the BE stream merge
As done for format and channels, add the possibility to merge
the backend rates on the frontend rates.

This useful if the backend does not support all rates supported by the
frontend, or if several backends (cpu and codecs) with different
capabilities are connected to the same frontend.

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-11 11:58:41 +01:00
Jerome Brunet 435ffb76f8
ASoC: dpcm: rework runtime stream merge
The goal of this patch is to simplify a bit dpcm runtime stream merge
by removing several local variables.

ATM, merge functions return the BE 'filter' values which should then be
filtered against the FE stream values. This create a lot of local
variable and unnecessary init of min and max.

Instead of this, we can pass the FE stream values directly and let the
BE filtering functions perform the merge 'in-place'

Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-11 11:58:37 +01:00
Timo Wischer ff2d6acdf6 ALSA: pcm: Fix snd_interval_refine first/last with open min/max
Without this commit the following intervals [x y), (x y) were be
replaced to (y-1 y) by snd_interval_refine_last(). This was also done
if y-1 is part of the previous interval.
With this changes it will be replaced with [y-1 y) in case of y-1 is
part of the previous interval. A similar behavior will be used for
snd_interval_refine_first().

This commit adapts the changes for alsa-lib of commit
9bb985c ("pcm: snd_interval_refine_first/last: exclude value only if
also excluded before")

Signed-off-by: Timo Wischer <twischer@de.adit-jv.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-11 08:49:59 +02:00
Hans de Goede caed9d636e
ASoC: Intel: bytcr_rt5651: Reporting button presses
Enable reporting of button presses now that the codec driver recently has
gotten support for this.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:28 +01:00
Hans de Goede b91f432cbc
ASoC: Intel: bytcr_rt5651: Disable jack-detect over suspend/resume
Disable jack-detection and thus the codec IRQ over suspend/resume.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:27 +01:00
Hans de Goede df1569f200
ASoC: rt5651: Add button press support
Enable button press detection for headsets by using the ovcd IRQ to get
notified of button presses.

This is modelled after (almost exactly copied from) the button press code
for the rt5640 which has identical ovcd hardware.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:26 +01:00
Hans de Goede 34c906ddac
ASoC: rt5651: Allow disabling jack-detect by calling set_jack(NULL)
Allow the machine driver to disable jack-detect over a suspend/resume by
calling snd_soc_component_set_jack(NULL).

Note this renames rt5651_set_jack, where all the jack-enable work was done
to rt5651_enable_jack_detect. This function can now no longer fail as it
does not request the IRQ anymore. It can still be passed an invalid jack
source, but that should never happen, so this is now logged and treated as
no jack source.

Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:25 +01:00
Hans de Goede 8d2d7bcdc1
ASoC: rt5651: Fix workqueue cancel vs irq free race on remove
On removal we must free the IRQ *before* cancelling the jack-detect work,
so that the jack-detect work cannot be rescheduled by the IRQ.

Before this commit we were cancelling the jack-detect work from the
driver remove callback, while relying on devm to free the IRQ, which
happens after the remove callback.

This is the wrong order. This commit uses a devm-action to register
a devm callback which cancels the work, before requesting the IRQ
(devm tears things down in reverse order). This also allows us to
remove the now empty remove driver callback.

Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:24 +01:00
Hans de Goede 5f6fb23d2e
ASoC: Intel: bytcr_rt5651: Add support for externar amplifier enable GPIO
The rt5651 does not have a built-in speaker amplifier, so it is often
used together with an external amplifier. On Cherry Trail boards this
external amplifier's enable pin is driven through a GPIO, which is
given as the first GPIO in the ACPI resources of the codec fwnode.

This commit adds support to the bytcr_rt5651 for this GPIO, fixing
the speaker not working on CHT devices with a rt5651 codec.

Cc: Carlo Caione <carlo@endlessm.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:23 +01:00
Hans de Goede 2c375204bf
ASoC: Intel: bytcr_rt5651: Move getting of codec_dev into probe()
Move the getting of the codec_dev, to add device-props to it, out of
byt_rt5651_add_codec_device_props() and into its caller,
snd_byt_rt5651_mc_probe().

This is a preparation patch for adding support for an external amplifier
enable GPIO, which requires further accesses to the codec_dev.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:17 +01:00
Hans de Goede fbea16dbc0
ASoC: Intel: bytcr_rt5651: Remove is_valleyview helper
Remove is_valleyview helper, this is not necessary, we can simply call
x86_match_cpu() directly instead.

Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:13 +01:00
Hans de Goede 81583afe79
ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix2 8 tablet
Add a quirk for the Lenovo Miix2 8 tablet, this tablet uses a digital
mic on DMIC1 and has a mono-speaker. The jack-detect uses the default
settings..

Reported-and-tested-by: russianneuromancer@ya.ru
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-10 18:49:10 +01:00
Marek Szyprowski d8095f94e1 dmaengine: add support for reporting pause and resume separately
'cmd_pause' DMA channel capability means that respective DMA engine
supports both pausing and resuming given DMA channel. However, in some
cases it is important to know if DMA channel can be paused without the
need to resume it. This is a typical requirement for proper residue
reading on transfer timeout in UART drivers. There are also some DMA
engines with limited hardware, which doesn't really support resuming.

Reporting pause and resume capabilities separately allows UART drivers to
properly check for the really required capabilities and operate in DMA
mode also in systems with limited DMA hardware. On the other hand drivers,
which rely on full channel suspend/resume support, should now check for
both 'pause' and 'resume' features.

Existing clients of dma_get_slave_caps() have been checked and the only
driver which rely on proper channel resuming is soc-generic-dmaengine-pcm
driver, which has been updated to check the newly added capability.
Existing 'cmd_pause' now only indicates that DMA engine support pausing
given DMA channel.

Signed-off-by: Marek Szyprowski <m.szyprowski@samsung.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
2018-07-09 22:59:04 +05:30
Lars-Peter Clausen 5bea327962
ASoC: adau171x1: Connect playback DAI to the DSP
The playback DAI is connected to the DSP and the DSP might be sourcing
signals from the playback stream. Add a DAPM route between the two to make
sure that the playback DAI is powered up, when the DSP is active.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Alexandru Ardelean <alexandru.ardelean@analog.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-09 12:29:03 +01:00
Arnd Bergmann 9d1310daed
ASoC: pxa: make SND_PXA_SOC_SSP depend on PLAT_PXA
For the moment, we can't enable CONFIG_SND_PXA_SOC_SSP unless we are
building for ARM PXA or MMP:

WARNING: unmet direct dependencies detected for PXA_SSP
  Depends on [n]: PLAT_PXA [=n]
  Selected by [y]:
  - SND_PXA_SOC_SSP [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y]

This adds an explicit dependency for it.

Fixes: 0a94cf3457 ("ASoC: pxa: make SND_PXA2XX_SOC_I2S selectable")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-09 12:27:16 +01:00
benjamin.gaignard@linaro.org 8db339d667
ASoC: stm32: replace "%p" with "%pK"
The format specifier "%p" can leak kernel addresses.
Use "%pK" instead.

Signed-off-by: Benjamin Gaignard <benjamin.gaignard@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-09 12:15:30 +01:00
Liam Girdwood d64c5cf8e8 ALSA: pcm: Allow drivers to set R/W wait time.
Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO.
This needs to be configurable for modern hardware like DSPs where no
pointer update in milliseconds can indicate terminal DSP errors.

Add a substream variable to set the wait time in ms. This allows userspace
and drivers to recover more quickly from terminal DSP errors.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-06 15:00:25 +02:00
Hui Wang c6b17f1020 ALSA: hda/realtek - two more lenovo models need fixup of MIC_LOCATION
We have two new lenovo desktop models which need to apply the fixup of
ALC294_FIXUP_LENOVO_MIC_LOCATION, and they have the same pin cfg as
the machine with subsystem id:0x17aa3136, now use the pincfg table
to apply the fixup for them.

Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2018-07-06 12:49:10 +02:00
Gustavo A. R. Silva f7ddff54d0
ASoC: nau8824: use 64-bit arithmetic instead of 32-bit
Add suffix ULL to constant 256 in order to give the compiler complete
information about the proper arithmetic to use.

Notice that such constant is used in a context that expects an
expression of type u64 (64 bits, unsigned) and the following
expression is currently being evaluated using 32-bit arithmetic:

256 * fs * 2 * mclk_src_scaling[i].param

Addresses-Coverity-ID: 1432039 ("Unintentional integer overflow")
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-05 16:09:11 +01:00