The AD1934 codec has no ADC feature. Hence it register mapping is slightly
different from the register mapping of other members of the AD193x family.
Some ASoC controls and widgets are related to the DAC feature so are not
relevant in the case of an AD1934 codec.
Signed-off-by: Cyrille Pitchen <cyrille.pitchen@atmel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, ESAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power
source of modules during system suspend and resume procedure. Thus,
SSI needs to save all the values of registers before the system
suspend and restore them after the system resume.
The register SFCSR is volatile, but some bits in it need to be
recovered after suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SPDIF needs to save
all the values of registers before the system suspend and restore them after
the system resume.
The SRPC register should be volatile, LOCK bit is set by the hardware.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume
be called in suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dw i2s controller can work in slave mode, codec being master.
dw i2s is made to support master/slave operation, by reading dwc
register.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
if the stream is decoupled and both link and host are used, while
releasing the stream, need to check if link and host stream are
not in use. This patch adds fix to check if the host/link stream
is in used before coupling it back when releasing the stream.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bits in LOSIDV need to be set to map the stream id for specific link.
Fixing this by setting the required bits in the register.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently if bulk_enable() of supplies fails, the code still goes on
to try and put the device into active state, and set expected IO
voltage of the device. This doesn't really make sense so code now
returns on bulk_enable() failure, with an error message. Also,
to help with debug, failure to get supplies also provides an error
message.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When OSS emulation is loaded on ISA SB AWE32 chip, we get now kernel
warnings like:
WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80()
sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0'
It's because both emux synth and opl3 drivers try to register their
OSS device object with the same static index number 0. This hasn't
been a big problem until the recent rewrite of device management code
(that exposes sysfs at the same time), but it's been an obvious bug.
This patch works around it just by using a different index number of
emux synth object. There can be a more elegant way to fix, but it's
enough for now, as this code won't be touched so often, in anyway.
Reported-and-tested-by: Michael Shell <list1@michaelshell.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the codec reset setting in the shutdown function to prevent
the weird sound of the headphone happened by rebooting.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent widget power saving introduced some unavoidable click
noises on old IDT 92HD73xx chips while it still seems working on the
compatible new chips. In the bugzilla, we tried lots of tests and
workarounds, but they didn't help much. So, let's disable the feature
for these specific chips as the least (but safest) fix.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Its a bit odd that debugfs_create_bool() takes 'u32 *' as an argument,
when all it needs is a boolean pointer.
It would be better to update this API to make it accept 'bool *'
instead, as that will make it more consistent and often more convenient.
Over that bool takes just a byte.
That required updates to all user sites as well, in the same commit
updating the API. regmap core was also using
debugfs_{read|write}_file_bool(), directly and variable types were
updated for that to be bool as well.
Signed-off-by: Viresh Kumar <viresh.kumar@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The MacBookPro 12,1 has the same setup as the 11 for controlling the
status of the optical audio light. Simply apply the existing workaround
to the subsystem ID for the 12,1.
[sorted the fixup entry by tiwai]
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105401
Signed-off-by: John Flatness <john@zerocrates.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Much like all the other Lenovo laptops, add a quirk to make
sound work with docking.
Reported-and-tested-by: lacknerflo@gmail.com
Signed-off-by: Laura Abbott <labbott@fedoraproject.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec supports 4 channel recording with TDM on AIF1.
This patch modifies the DAI capability to allow it.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for the DA7219 audio codec with built-in advanced
accessory detect features.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Designware I2S uses tx empty and rx available signals as the DMA
handshaking signals. during music playing, if XRUN occurs,
i2s_stop() function will be executed and both tx and rx irq are
masked, when music continues to be played, i2s_start() is executed
but both tx and rx irq are not unmasked which cause I2S stop
sending DMA handshaking signal to DMA controller, and it finally
causes music playing will be stopped once XRUN occurs for the first
time.
[On list discussion suggests this may be partly a race condition on slow
systems -- broonie]
Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The struct snd_soc_tplg_pcm_dai is renamed to snd_soc_tplg_pcm.
This struct will now be used to handle data related to PCMs
(FE DAI & DAI links). It's not for BE, because BE DAI mappings will be
provided by ACPI/FDT data.
Remove the unused struct snd_soc_tplg_pcm_cfg_caps. We are using
snd_soc_tplg_stream and snd_soc_stream_caps instead.
Bump ABI version to 4.
Signed-off-by: Vedang Patel <vedang.patel@intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds convenience defines for declaring a gain control that
has an input mux. These blocks are functionally equivalent to
the existing mixer blocks but can only have a single input
active at once.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds hwdep interface so as the other IEEE 1394 sound devices
has.
This interface is designed for mixer/control applications. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds PCM functionality to transmit/receive PCM samples.
When one of PCM substreams are running or external clock source is
selected, current sampling rate is used. Else, the sampling rate is
changed as an userspace application requests.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds streaming functionality for both direction. To utilize
the sequence of the number of data blocks in packets, full duplex with
synchronization is applied.
Besides, TASCAM FireWire series allows drivers to decide which PCM data
channels are enabled. For convenience, this driver always enable whole the
data channels.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series uses non-blocking transmission for AMDTP packet
streaming, while the format of data blocks is unique.
The CIP headers includes specific value in FMT field and no SYT
information.
In transmitted packets, the first data channel represents event counter,
and the last data channel has status and control information. The rest
has 24bit PCM samples with right padding.
In received packets, all of data channels include 16, 24, 32bit PCM
samples. There's no other kind of information.
This commit adds support for this protocol. For convenience, the size of
PCM samples in outgoing packet is limited by 16 and 24bit. The status and
control information will be supported in future commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series has certain registers for firmware information.
This commit adds proc node to show the information.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series doesn't tell drivers their capabilities, thus
the drivers should have model-dependent parameters and apply it to
detected devices.
This commit adds a structure to represent such parameters.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for TASCAM FireWire series. In this commit,
this driver just creates/removes card instance according to bus event.
More functionalities will be added in following commits.
TASCAM FireWire series consists of:
* PDI 1394P23 for IEEE 1394 PHY layer
* PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
* XILINX XC9536XL
* XILINX Spartan-II XC2S100
* ATMEL AT91M42800A
Ilya Zimnovich had investigated TASCAM FireWire series in 2011, and
discover some features of his FW-1804. You can see a part of his research
in FFADO project.
http://subversion.ffado.org/wiki/Tascam
A part of my work are based on Ilya's investigation, while this series
doesn't support the FW-1804, because of a lack of config ROM
information and its protocol detail, especially for PCM channels.
I observed that FW-1884 and FW-1082 don't work properly with 1394 OHCI
controller based on VT6315. The controller can actually communicate packets
to these models, while these models generate no sounds. It may be due to
the PHY/LINK layer issues. Using 1394 OHCI controller produced by the other
vendors such as Texas Instruments may work. Or adding another node on the
bus.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
from Designware I2S datasheet, tx/rx XRUN irq is cleared by
reading register TOR/ROR, rather than by writing into them.
Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The current code writes a set of registers that are reserved on the
tlc320aic3104. The change skips those registers for that IC.
Signed-off-by: Rick Mann <rmann@latencyzero.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes the old PXA DMA API usage and switches over to
generic functions provided by snd-soc-dmaengine-pcm.
More cleanups may be done on top of this, and some function stubs can
now be removed completetly. However, the intention here was to keep
the transition as small as possible.
This was tested on the mioa701 pxa27x board.
Signed-off-by: Daniel Mack <zonque@gmail.com>
[trivial change from mmp-dma to pxa-dma]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' into asoc-pxa
ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
The SND_PCM_RATE_KNOT covers all the rate settings, even though some that
we don't support, while we also list all the rate we support. Simply remove
it.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current code, disregarding the clk_set_rate error code, was always
returning -EINVAL. Fix that and return the code in order to have more clue
about what's going on.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
pm_runtime_enable is called in probe to enable runtime PM
for wm8962 codec, but pm_runtime_disable isn't called in remove
callback, nor is called in error path if probe fails after runtime
PM is enabled, this causes unbalanced pm_runtime_enable.
This patch Adds pm_runtime_disable in remove callback and error path,
to balance pm_runtime_enable.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Digi 002/003 family uses asynchronous transaction for messaging.
The address to transmit this message is stored on a certain register.
This commit allocates a range of address on OHCI 1394 host controller
to handle the messaging. As long as I know, the purpose of this message
seems to notify lost of synchronization. While, the meaning of content
of the message is not clear.
Actual examples of this messaging:
* When clock source is set as internal:
- 0x00007051
- 0x00007052
- 0x00007054
- 0x00007057
- 0x00007058
* When clock source is set as somewhat external:
- 0x00009000
- 0x00009010
- 0x00009020
- 0x00009021
- 0x00009022
The lost often occurs when using internal clock source. In this case,
users hear sounds with quite short gap every several minutes. In fact,
the lost is recovered temporarily.
When using with external clock source, the lost seems not to occur. The
mechanism is not clear yet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds hwdep interface so as the other sound drivers for units
on IEEE 1394 bus have.
This interface is designed for mixer/control applications. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds PCM functionality to transmit/receive PCM samples.
Any PCM substreams are jointed because incoming/outgoing AMDTP streams
are bound. When one of PCM substream is running or external clock source
is selected, current sampling rate is used. Else, the sampling rate is
changed as an userspace application requests.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds proc node to show current clock status for debugging.
As long as testing Digi 002 rack, registers can show local clock rate,
local clock source. When external clock input such as S/PDIF is
connected, the registers show the detection and external clock rate.
Additionally, the registers show the mode of optical digital input
interface. Although, a tester with Digi 003 rack reports this makes no
sense. Further investigation is required for Digi 003 series.
Besides, in Digi 002 rack, the S/PDIF format must be IEC 60958-4,
so-called professional.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to manage streaming.
The streaming is not controlled by CMP in IEC 61883-6. It's controlled by
IEEE 1394 write transaction to certain addresses.
Several clock sources are available, while there're no differences about
packet transmission. The value of SYT field in transmitted packets is
always zero. Thus, streams in both direction don't build synchronization.
And the device always requires received packets to transmit packets. This
driver keeps to transfer outgoing stream even if they're not required.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 002/003 family uses its own format for data blocks. The format is
quite similar to AM824 in IEC 61883-6, while there're some differences:
* The Valid Bit Length (VBL) code is always 0x40 in Multi-bit Linear Audio
(MBLA) data channel.
* The first data channel includes MIDI messages, against IEC 61883-6
recommendation.
* The Counter field is always zero in MIDI conformant data channel.
* Sequence multiplexing in IEC 61883-6 is not applied to the MIDI
conformant data channel.
* PCM samples are scrambled in received AMDTP packets. We call the way
as Double-Oh-Three (DOT). The algorithm was discovered by
Robin Gareus and Damien Zammit in 2012.
This commit adds data processing layer to satisfy these differences.
There's a quirk about transmission mode for received packets. When this
driver applies non-blocking mode to outgoing packets with isochronous
channel 2 or more, after 15 to 20 seconds since playbacking, any PCM
samples causes noisy sound on the device. With isochronous channel 0 or 1,
this doesn't occur. As long as I investigated, this quirk is not observed
when applying blocking mode to the received packets.
This driver applies blocking mode to outgoing packets, while non-blocking
mode to incoming packgets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for Digidesign 002/003 family. This commit
just creates/removes card instance according to bus event. More functions
will be added in following commits.
Digidesign 002/003 family consists of:
* Agere FW802B for IEEE 1394 PHY layer
* PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
* ALTERA ACEX EP1K50 for IEC 61883 layer and DSP controller
* ADSP-21065L for signal processing
[minor cleanup using skip_spaces() by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another attempt at drm-misc for 4.4 ...
- better atomic helpers for runtime pm drivers
- atomic fbdev
- dp aux i2c STATUS_UPDATE handling (for short i2c replies from the sink)
- bunch of constify patches
- inital kerneldoc for vga switcheroo
- some vblank code cleanups from Ville and Thierry
- various polish all over
* tag 'topic/drm-misc-2015-09-25' of git://anongit.freedesktop.org/drm-intel: (57 commits)
drm/irq: Add drm_crtc_vblank_count_and_time()
drm/irq: Rename drm_crtc -> crtc
drm: drm_atomic_crtc_get_property should be static
drm/gma500: Remove DP_LINK_STATUS_SIZE redefinition
vga_switcheroo: Set active attribute to false for audio clients
drm/core: Preserve the fb id on close.
drm/core: Preserve the framebuffer after removing it.
drm: Use vblank timestamps to guesstimate how many vblanks were missed
drm: store_vblank() is never called with NULL timestamp
drm: Clean up drm_calc_vbltimestamp_from_scanoutpos() vbl_status
drm: Limit the number of .get_vblank_counter() retries
drm: Pass flags to drm_update_vblank_count()
drm/i915: Fix vblank count variable types
drm: Kill pixeldur_ns
drm: Stop using linedur_ns and pixeldur_ns for vblank timestamps
drm: Move timestamping constants into drm_vblank_crtc
drm/fbdev: Update legacy plane->fb refcounting for atomic restore
drm: fix kernel-doc warnings in drm_crtc.h
vga_switcheroo: Sort headers alphabetically
drm: Spell vga_switcheroo consistently
...
SND_SOC_DAIFMT_{IB|NB}_{IF|NF} are defined as inverting or not BCLK or
FRM relatively to what is standard for the specified DAI hardware audio
format. Consequently, the absolute polarities of these signals cannot be
derived only from these settings as this driver did. The format has to
be taken into account too.
This fixes inverted left/right channels in I²S mode.
Signed-off-by: Benoît Thébaudeau <benoit@wsystem.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_release_substream() always calls hw_free op when the stream
was opened. This is superfluous in most cases because it's been
already released via explicit hw_free ioctl. Although this double
call is usually OK as this callback should be written to be called
multiple times, it's better to avoid superfluous calls.
Reported-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja Kp <jeeja.kp@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit moves the codes related to data block processing from packet
streaming layer to AM824 layer.
Each driver initializes amdtp stream structure for AM824 data block by
calling amdtp_am824_init(). Then, a memory block is allocated for AM824
specific structure. This memory block is released by calling
amdtp_stream_destroy().
When setting streaming parameters, it calls amdtp_am824_set_parameters().
When starting packet streaming, it calls amdtp_stream_start(). When
stopping packet streaming, it calls amdtp_stream_stop().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit renames some macros just related to AM824 format. In later
commit, they're moved to AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting the format of PCM substream to AMDTP stream structure is important
to set a handler to copy actual PCM samples between buffers. The
processing should be in data block processing layer because essentially
it has no relationship to packet streaming.
This commit renames PCM format setting function to prepare for integrating
AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, MIDI messages are transferred in MIDI conformant data
channel. Essentially, packet streaming layer is not responsible for MIDI
functionality.
This commit moves MIDI trigger helper function from the layer to AM824
layer. The rest of codes related to MIDI functionality will be moved in
later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, several types of data are available in AM824 format. The
data is transferred in each data channel. The position of data channel in
data block differs depending on model.
Current implementation has an array to map the index of data channel in an
data block to the position of actual data channel. The implementation
allows each driver to access the mapping directly.
In later commit, the mapping is in specific structure pushed into an
opaque pointer. Helper functions are required.
This commit adds the helper functions for this purpose. In IEC 61883-6,
AM824 format supports many data types, while this specification easily
causes over-engineering. Current AM824 implementation is allowed to handle
two types of data, Multi Bit Linear Audio data (=PCM samples) and MIDI
conformant data (=MIDI messages).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, PCM frames are transferred in Multi Bit Linear Audio data
channel. The data channel transfers 16/20/24 bit PCM samples. Thus, PCM
substream has a constrain about it.
This commit moves codes related to the constraint from packet streaming
layer to AM824 data block processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The value of FDF field in CIP header is protocol-dependent. Thus, it's
better to allow data block processing layer to decide the value in any
timing.
In AM824 data format, the value of FDF field in CIP header indicates
N-flag and Nominal Sampling Frequency Code (sfc). The N-flag is for
switching 'Clock-based rate control mode' and 'Command-based rate control
mode'. In our implementation, 'Clock-based rate control mode' is just
supported. Therefore, When sampling transfer frequency is decided, then
the FDF can be set.
This commit replaces 'amdtp_stream_set_parameters' with
'amdtp_am824_set_parameters' to set the FDF. This is the same timing
to decide the ration between the number of data blocks and the number of
PCM frames.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds data block processing layer for AM824 format. The new
layer initializes streaming layer with its value for fmt field.
Currently, most implementation of data block processing still remains
streaming layer. In later commits, these codes will be moved to the layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, data block processing layer will be newly added. This
layer will be named as 'amdtp-am824'.
This commit renames current amdtp file to amdtp-stream, to distinguish it
from the new layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some vendor specific protocol uses its own value for fmt/fdf fields in
CIP header.
This commit support to set arbitrary values for the fields.
In IEC 61883-6, NO-DATA code is defined for FDF field. A packet with this
code includes no data even if it includes some data blocks. This commit
still leaves a condition to handle this special packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM framework uses PCM buffer with a concept of 'period' to
synchronize userspace operations to hardware for nearly-realtime
processing. Each driver implements snd_pcm_period_elapsed() to tell across
of the period boundary to ALSA PCM middleware. To call the function, some
drivers utilize hardware interrupt handlers, the others count handled PCM
frames.
Drivers for sound units on IEEE 1394 bus are the latter. They use two
buffers; PCM buffer and DMA buffer for IEEE 1394 isochronous packet. PCM
frames are copied between these two buffers and 'amdtp_stream' structure
counts the handled PCM frames. Then, snd_pcm_period_elapsed() is called if
required.
Essentially, packet streaming layer should not be responsible for PCM
frame processing. The PCM frame processing should be handled in each data
block processing layer as a result of handling data blocks. Although, PCM
frame counting is a common work for all of protocols which ALSA firewire
stack is going to support.
This commit adds two new helper functions as interfaces between packet
streaming layer to data block processing layer. In future, each data block
processing layer implements these functions. The packet streaming layer
calls data block processing layer per packet by calling the functions. The
data block processing layer processes data blocks and PCM frames, and
returns the number of processed PCM frames. Then the packet streaming layer
calculates handled PCM frames and calls snd_pcm_period_elapsed().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In future commit, interface between data block processing layer and packet
stream processing layer is defined. These two layers communicate the
number of data blocks and the number of PCM frames.
The data block processing layer has a responsibility for calculating the
number of PCM frames. Therefore, 'dual wire' of Dice quirk should be
handled in data block processing layer.
This commit adds a member of 'frame_multiplier'. This member represents
the ratio of the number of PCM frames against the number of data blocks.
Usually, the value of this member is 1, while it's 2 in Dice's 'dual wire'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, one data block represents one event. In ALSA, the event is
one PCM frame. Therefore, when processing one data block, current
implementation counts one PCM frame.
On the other hand, Dice platform has a quirk called as 'dual wire' at
higher sampling rate. In detail, see comment of commit 6eb6c81eee
("ALSA: dice: Split stream functionality into a file").
Currently, to handle this quirk, AMDTP stream structure has a
'double_pcm_frames' member. When this is enabled, two PCM frames are
counted. Each driver set this flag by accessing the structure member
directly.
In future commit, some members related to AM824 data block will be moved
to specific structure, to separate packet streaming layer and data block
processing layer. The access will be limited by opaque pointer.
For this reason, this commit adds an argument into
amdtp_stream_set_parameter() to set the flag.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, amdtp_stream_set_parameters() returns no error even if wrong
arguments are given. This is not good for streaming layer because drivers
can continue processing ignoring capability of streaming layer.
This commit changes this function to return error code.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, some members related to AM824 data format will be moved
from AMDTP stream structure to data block structure. This commit is a
preparation for it. Additionally, current layout of AMDTP stream structure
is a bit mess by several extensions. This commit also arranges the layout.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to verify that "value" is either zero or one, so we test if it
is greater than one. Unfortunately, this is a signed int so it could
also be negative. I think this is harmless but it introduces a static
checker warning. Let's make "value" unsigned.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit b4508d0f95 ("ASoC: db1200: Use static DAI format setup") switched
the db1200 driver over to using static DAI format setup instead of a
callback function. But the commit only added the dai_fmt field to one of
the three DAI links in the driver. This breaks audio on db1300 and db1550.
Add the two missing dai_fmt settings to fix the issue.
Fixes: b4508d0f95 ("ASoC: db1200: Use static DAI format setup")
Reported-by: Manuel Lauss <manuel.lauss@gmail.com>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
For display audio, call the sync_audio_rate callback function
to do the synchronization between gfx driver and audio driver.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
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Merge tag 'asoc-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.4
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
Lenovo Thinkpads with recent Realtek codecs seem suffering from click
noises at power transition since the introduction of widget power
saving in 4.1 kernel. Although this might be solved by some delays in
appropriate points, as a quick workaround, just disable the
power_save_node feature for now. The gain it gives is relatively
small, and this makes the situation back to pre 4.1 time.
This patch ended up with a bit more code changes than usual because
the existing fixup for Thinkpads is highly chained. Instead of adding
yet another chain, combine a few of them into a single fixup entry, as
a gratis cleanup.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
The active attribute in struct vga_switcheroo_client denotes whether
the outputs are currently switched to this client. The attribute is
only meaningful for vga clients. It is never used for audio clients.
The function vga_switcheroo_register_audio_client() misuses this
attribute to store whether the audio device is fully initialized.
Most likely there was a misunderstanding about the meaning of
"active" when this was added.
Comment from Takashi's review:
"Not really. The full initialization of audio was meant that the audio
is active indeed. Admittedly, though, the active flag for each audio
client doesn't play any role because the audio always follows the gfx
state changes, and the value passed there doesn't reflect the actual
state due to the later change. So, I agree with the removal of the
flag itself -- or let the audio active flag following the
corresponding gfx flag. The latter will make the proc output more
consistent while the former is certainly more reduction of code."
Set the active attribute to false for audio clients. Remove the
active parameter from vga_switcheroo_register_audio_client() and
its sole caller, hda_intel.c:register_vga_switcheroo().
vga_switcheroo_register_audio_client() was introduced by 3e9e63dbd3
("vga_switcheroo: Add the support for audio clients"). Its use in
hda_intel.c was introduced by a82d51ed24 ("ALSA: hda - Support
VGA-switcheroo").
v1.1: The changes above imply that in find_active_client() the call
to client_is_vga() is now superfluous. Drop it.
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Lukas Wunner <lukas@wunner.de>
[danvet: Add Takashi's clarification to the commit message.]
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Also move the include of sound/hda_verbs.h to rl6347a.h because it is used
in rl6347a.h.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Tegra HD-audio controller driver causes deadlocks when loaded as a
module since the driver invokes request_module() at binding with the
codec driver. This patch works around it by deferring the probe in a
work like Intel HD-audio controller driver does. Although hovering
the codec probe stuff into udev would be a better solution, it may
cause other regressions, so let's try this band-aid fix until the more
proper solution gets landed.
Reported-by: Thierry Reding <treding@nvidia.com>
Tested-by: Thierry Reding <treding@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As far as I can see, having an invalid ->tstamp_mode is harmless, but
adding a check silences a static checker warning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mackie Onyx Satellite has two usage; standalone and with base station.
These two modes has different stream formats. In standalone mode, rx data
block includes 2 Multi Bit Linear Audio (MBLA) data channels and tx data
block includes 2. With base station, they're 6 and 2.
Although, with base station, the actual tx packet include wrong value in
'dbs' field in its CIP header. This quirk causes packet streaming layer to
detect packet discontinuity and to stop PCM substream.
This is a response of 'single' subfunction 'extended stream format
information' command in AV/C Stream Format Information Specification 1.1.
It means that a data block in tx packets includes 2 MBLA data channels.
$ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffffffff
response: 000: 0c ff bf c0 01 00 00 00 ff 00 90 40 03 02 01 02
response: 010: 06
Current OXFW driver parses the response and detects stream formats
correctly.
$ cat /proc/asound/card1/firewire/formation
...
Output Stream from device:
Rate PCM MIDI
* 48000 2 0
44100 2 0
88200 2 0
96000 2 0
On the other hand, in actual tx CIP, the 'dbs' field has 6. But the number
of quadlets in CIP payload is not multiple of 6. Seeing the subtraction of
two successive payload quadlets, it should be multiple of 2.
payload CIP CIP
quadlets header0 header1
24 00060052 9002ffff
24 0006005e 9002ffff
26 0006006a 9002ffff
24 00060077 9002ffff
24 00060083 9002ffff
26 0006008f 9002ffff
24 0006009c 9002ffff
24 000600a8 9002ffff
26 000600b4 9002ffff
24 000600c1 9002ffff
This commit adds support for this quirk to OXFW driver, by using
CIP_WRONG_DBS flag. When this flag is set, packet streaming layer uses
the value of its 'data_block_quadlets' member instead of the value in CIP
header. This value is already set by OXFW driver and no discontinuity is
detected.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PCM receive and transmit DMA requestor lines were reverted, breaking the
PCM playback interface for PXA platforms using the sound/soc/ variant
instead of the sound/arm variant.
The commit below shows the inversion in the requestor lines.
Fixes: d65a14587a ("ASoC: pxa: use snd_dmaengine_dai_dma_data")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Broadwell can not triger the IRQ falling and rising simultaneously,
so it can not detect jack-in and jack-out simultaneously.
We add a flag "jd_invert" to platform data. If this flag is set,
codec IRQ will be set to invert that forces IRQ as pulse when jack-in
and jack-out.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add DMI data for Buddy project.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current code incorrectly treats dai format for AC97 as bit mask
whereas it's actually an integer value. This causes DAI formats
other than AC97 (e.g. DSP_B) to trigger AC97 related code,
which is incorrect and breaks functionality. This patch fixes
the code to correctly compare values to determine AC97 or not.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the capability to use multiple codecs on the same DAI linke where
one codec is used for playback and another one is used for capture.
Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS
microphone for capture.
No verification is made that the playback and capture codec setups are
compatible in terms of number of TDM slots (where applicable).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The hdmi stub codec has not been used since refactoring of OMAP HDMI
audio support.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to create CPU DAI for each endpoint instance. For this we
should have one DMIC DAI, one HDA DAI and SSP DAI. Thus, DMIC23,
HDA-SPK/AMIC was not required so this patch removes them
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the error path so that we can free the allocated memory on the error
path instead of releasing them individually on each error.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have requested for the firmware but we have missed releasing it both
on success and on error path.
While checking the code it turned out that the requested firmware is not
even used. More over the same firmware is being loaded by
wm0010_stage2_load().
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the max register value of mic boost pga should be 3.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm8962 can't support 64k sample rate. When playing a 64KHz wave file,
'Unsupported rate 64000Hz' will be prompted.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make SND_SOC_ROCKCHIP_MAX98090 and SND_SOC_ROCKCHIP_RT5645 depend on
CLKDEV_LOOKUP to fix below build warning:
warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects
SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K &&
!UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP)
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The '\n' at the end of the format string is not needed. It adds an extra
line break when doing
cat /proc/interrupts
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
CC: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The slot_width is for essentially same thing. Instead of storing
bclk_lrclk_ratio, just store the slot_width. Comments has been updated
accordingly and some variable names changed to more descriptive.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Apply PTR_ERR to the value that was recently assigned.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,y;
@@
if (IS_ERR(x) || ...) {
... when any
when != IS_ERR(...)
(
PTR_ERR(x)
|
* PTR_ERR(y)
)
... when any
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Store return value of of_get_property() to a pointer of __be32 type as
device tree has big endian type. This fixes a sparse warning couple of
lines later when be32_to_cpup() is used to convert from big endian to
cpu endian.
The whole conversion is not really necessary, as we are only checking
if the value is zero or not, but I wanted to add it to remind in the
future that the data has to be converted before use. Compiler should
optimize the unnecessary operations away.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Earlier revisions of the wm5110/8280 silicon require a slightly more
complex procedure to enable analogue inputs. This patch adds this into
the driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We will occasionally require to take different action based on if an
input is analog or digital so add a helper function to return if an
input is analog.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the addition of the WILL_PMU / WILL_PMD several of the switches in
arizona.c no longer cover all cases or have a default case. Whilst this
isn't causing any problems in the interests of robustness add default
cases.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous fix of pxa library support, which was introduced to fix the
library dependency, broke the previous SoC behavior, where a machine
code binding pxa2xx-ac97 with a coded relied on :
- sound/soc/pxa/pxa2xx-ac97.c
- sound/soc/codecs/XXX.c
For example, the mioa701_wm9713.c machine code is currently broken. The
"select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for
compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared
in sound/arm/Kconfig and sound/soc/pxa/Kconfig.
Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct
pxa2xx-ac97 compilation.
Fixes: 846172dfe3 ("ASoC: fix SND_PXA2XX_LIB Kconfig warning")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") added AUDIO_CLKOUT
support for ADG. But single AUDIO_CLKOUT needs clkout_name[CLKOUT],
not clkout_name[i]. Kernel will have NULL pointer access without this
patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
71a0138ab("ASoC: ak4642: enable to use MCKO as fixed rate output
pin on DT") added new FS() macro, but x86 already has it in
arch/x86/include/uapi/asm/ptrace-abi.h
This patch exchange FS() to FSs() to avoid redefinition warning
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") uses
of_clk_add_provider() which is requesting struct clk_onecell_data.
But it is COMMON_CLK feature. SND_SOC_RCAR depends on COMMON_CLK
This patch also solved compile error of 7486d80f7("ASoC: rsnd: remove
unneeded sh_clk header")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The sun4i, sun5i and sun7i SoC families have a built-in codec, capable
of both audio capture and playback.
While this is called a codec by Allwinner, it really is an in-SoC
combination of a codec and a DAI, with its own DAC/ADC and amplifiers
in a single memory-mapped controller.
The capture part has been left out for now, and will be added eventually.
Signed-off-by: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The function get_current_pipe_id() was not being used.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds DT binding for explicitly choosing a tdm mask for DAI and uses it
in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been
changed.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implements set_tdm_slot() callback for mcasp. Channel constraints are
updated according to the configured tdm mask and slots each time
set_tdm_slot() is called. The special case when slot width is set to
zero is allowed and it means that slot width is the same as the sample
width.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Before this patch the set_tdm_slots() callback did not store the value
of slot width anywhere. The tdm support only worked if selected slot
width was equal to the sample width. With this patch all sample widths
that fit into the slot width are supported. There unused bits are
filled unnecessarily in the capture direction, but the other end of
the i2s bus should be able to ignore them.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has AUDIO_CLKOUT (in Gen1/Gen2) AUDIO_CLKOUT1/2/3 (in Gen3)
This patch support these patches as clock provider.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
undefined clock is not error. Accept such case. And this is prepare
for clock out support in the same time.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It didn't have "\n", and indicated different data
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ADG clock calculation needs ADG and SSI settings.
Thus, SSI side clock request function depends on ADG settings.
After reconsideration, we can close this method inside ADG.
This function uses new method. And it becomes preparation for
AUDIO_CLKOUT support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ADG is special IP since it doesn't have MSTP. And now, ADG has common
mod base register access. We can now setup ADG initial setting when
probe timing. It is needed if sound card is based on AUIDO_CLK which
is used as Master clock.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has SSI/SRC/DVC/MIX/ADG modules, and these have original
register mapping. Thus this driver is using regmap field, and each module
is using it based on each module ID.
Sometimes, each module needs other module to controlling. but current each
function is using just "mod" as parameter name. This is confusable.
For example, if SSI0 and SRC2 are connected, and if SRC module function
has bug of module access, and if it needs to control connected SSI,
SRC function will access to SSI2 (It should access to SSI0, but it uses
SRC's ID 2). This is easy to happen in current driver style.
To avoid this kind of confusable trouble, this patch adds module confirm
macro for debug purpose.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has ADG IP, but it is special device.
(It is clock generater, and it doesn't need MSTP)
Thus, ADG didn't use mod base common method on rsnd driver.
But it can be confusable point. Let's use common method
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound driver has SSI/SRC/DVC/CTU/MIX, and these are controlled
as modules. And these module are member of each modules's private data.
It used own method to get module pointer, but Let's use common method
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound Gen3 is updated version of Gen2. We need to update
driver for it, but basically it should works as Gen2 compatible.
This is initial support for Gen3. Gen3 specific feature will be
incrementally added in the future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sh_clk header is not needed, and it will create confusion.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_src_pcm_new() is used only from Gen2. make it clear in function name,
and remove unneeded Gen1 check.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_dma_to_xxx() macro should exist in same place
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ak4642 chip can output clock via MCKO pin as audio reference clock.
This patch supports it on DT
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The TX and RX direction share the same bit clock and frame sync, so
the samplerate must be the same to both directions.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
These platform drivers have a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix below build warning:
sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe':
sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable]
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In capture, there is chance that hw_ptr reported at IRQ is
a little smaller than period_size due to internal AFE buffer.
In the case of ping-pong buffer:
|xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------|
hw_ptr < period_size
This available buffer will not be read since its size is smaller than
avail_min (which is period_size by default), and read thread continues
to sleep. If the next hw_ptr is just a little larger than buffer_size,
overrun occurs. One more period can hold the possible unread buffer.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value
for the refining should meet this requirement as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This structure was added by 4d96eb255c ('ALSA: pcm_lib - add possibility
to log last 10 DMA ring buffer positions') to store PCM pointers
information of latest 10 pointer movements (=XRUN_LOG_CNT). When
CONFIG_SND_PCM_XRUN_DEBUG is configured, 'struct snd_pcm_runtime' has
'hwptr_log' member with a pointer to the structure. When calling
xrun_log() in pcm_lib.c, the structure was allocated to the pointer.
When calling snd_pcm_detach_substream() in pcm.c, the allocated pointer
is released.
In f5914908a5 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints'),
the pointer logging is replaced with using Linux Kernel Tracepoints. The
structure was also removed, while it's just declared. The member and kfree
still remains.
This commit removes the member and related codes. I think this was
overlooked because it brings no errors/warnings to C compilers.
Fixes: f5914908a5 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A collection of small fixes since the last update: the HD-audio
quirks as usual with a USB-audio fix and a trivial fix for the
old sparc driver.
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Merge tag 'sound-fix-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes since the last update: the HD-audio quirks
as usual with a USB-audio fix and a trivial fix for the old sparc
driver"
* tag 'sound-fix-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: usb-audio: Change internal PCM order
ALSA: hda - Fix white noise on Dell M3800
ALSA: hda - Use ALC880_FIXUP_FUJITSU for FSC Amilo M1437
ALSA: hda - Enable headphone jack detect on old Fujitsu laptops
ALSA: sparc: amd7930: Fix module autoload for OF platform driver
ALSA: hda - Add some FIXUP quirks for white noise on Dell laptop.
If the size of the firmware is less than expected size then we are
exiting with the error code but we missed releasing the firmware.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input PGAs have a gain range from -17.25dB to +30dB in 0.75dB steps.
The boost stage can provide additional gain. For line inputs, -12dB to
+6dB gain is available on the boost mixer. For micphone inputs, it can
provide up to +29dB additional gain from the microphone PGA.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The min gain is the corresponding gain value when the register value is 0
instead of 1, just correct it.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch removes the incorrect settings to avoid the pop sound in the
first playback with headphone after boot.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Incase of an unknown event we were directly returning but we missed
freeing params.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add check on of_property_read to return error when
DT required property is not defined.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the SSP port settings are being clobbered as part of the DSP
RTD3 restore logic. make sure we save the correct params and restore them
at resume. The FW sadly does not save SSP settings as part of the PM
context.
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.
This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.
Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M3800 is very minor workstation variant of the XPS 15 which has
already been patched for this issue. I figured it's probably more
important for this version of the laptop to be patched than the
regular XPS as Dell sells is pre-configured with Ubuntu to be used as
a Linux workstation. I have tested the patch on my the hardware on
Linux 4.2.0.
Signed-off-by: Niranjan Sivakumar <ns253@cornell.edu>
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It turned out that the machine has a bass speaker, so take a correct
fixup entry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to the bug report, FSC Amilo laptops with ALC880 can detect
the headphone jack but currently the driver disables it. It's partly
intentionally, as non-working jack detect was reported in the past.
Let's enable now.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are little changes in core part, but lots of development are
found in drivers, especially ASoC. The diffstat shows regmap-
related changes for a slight API additions / changes, and that's all.
Looking at the code size statistics, the most significant addition
is for Intel Skylake. (Note that SKL support is still underway, the
codec driver is missing.) Also STI controller driver is a major
addition as well as a few new codec drivers.
In HD-audio side, there are fewer changes than the past. The
noticeable change is the support of ELD notification from i915
graphics driver. Thus this pull request carries a few changes in
drm/i915.
Other than that, USB-audio got a rewrite of runtime PM code. It
was initiated by lockdep warning, but resulted in a good cleanup in
the end.
Below are the highlights:
Common:
- Factoring out of AC'97 reset code from ASoC into the core helper
- A few regmap API extensions (in case it's not pulled yet)
ASoC:
- New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
RT298 and ST STI controllers
- Machine drivers for Rockchip systems with MAX98090 and RT5645 and
RT5650
- Initial driver support for Intel Skylake devices
- Lots of rsnd cleanup and enhancements
- A few DAPM fixes and cleanups
- A large number of cleanups in various drivers (conversion and
standardized to regmap, component) mostly by Lars-Peter and Axel
HD-audio:
- Extended HD-audio core for Intel Skylake controller support
- Quirks for Dell headsets, Alienware 15
- Clean up of pin-based quirk tables for Realtek codecs
- ELD notifier implenetation for Intel HDMI/DP
USB-audio:
- Refactor runtime PM code to make lockdep happier
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Merge tag 'sound-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"There are little changes in core part, but lots of development are
found in drivers, especially ASoC. The diffstat shows regmap-related
changes for a slight API additions / changes, and that's all.
Looking at the code size statistics, the most significant addition is
for Intel Skylake. (Note that SKL support is still underway, the
codec driver is missing.) Also STI controller driver is a major
addition as well as a few new codec drivers.
In HD-audio side, there are fewer changes than the past. The
noticeable change is the support of ELD notification from i915
graphics driver. Thus this pull request carries a few changes in
drm/i915.
Other than that, USB-audio got a rewrite of runtime PM code. It was
initiated by lockdep warning, but resulted in a good cleanup in the
end.
Below are the highlights:
Common:
- Factoring out of AC'97 reset code from ASoC into the core helper
- A few regmap API extensions (in case it's not pulled yet)
ASoC:
- New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
RT298 and ST STI controllers
- Machine drivers for Rockchip systems with MAX98090 and RT5645 and
RT5650
- Initial driver support for Intel Skylake devices
- Lots of rsnd cleanup and enhancements
- A few DAPM fixes and cleanups
- A large number of cleanups in various drivers (conversion and
standardized to regmap, component) mostly by Lars-Peter and Axel
HD-audio:
- Extended HD-audio core for Intel Skylake controller support
- Quirks for Dell headsets, Alienware 15
- Clean up of pin-based quirk tables for Realtek codecs
- ELD notifier implenetation for Intel HDMI/DP
USB-audio:
- Refactor runtime PM code to make lockdep happier"
* tag 'sound-4.3-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (411 commits)
drm/i915: Add locks around audio component bind/unbind
drm/i915: Drop port_mst_index parameter from pin/eld callback
ALSA: hda - Fix missing inline for dummy snd_hdac_set_codec_wakeup()
ALSA: hda - Wake the codec up on pin/ELD notify events
ALSA: hda - allow codecs to access the i915 pin/ELD callback
drm/i915: Call audio pin/ELD notify function
drm/i915: Add audio pin sense / ELD callback
ASoC: zx296702-i2s: Fix resource leak when unload module
ASoC: sti_uniperif: Ensure component is unregistered when unload module
ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource
ASoC: sh: dma-sh7760: Convert to devm_snd_soc_register_platform
ASoC: spear_pcm: Use devm_snd_dmaengine_pcm_register to fix resource leak
ALSA: fireworks/bebob/dice/oxfw: fix substreams counting at vmalloc failure
ASoC: Clean up docbook warnings
ASoC: txx9: Convert to devm_snd_soc_register_platform
ASoC: pxa: Convert to devm_snd_soc_register_platform
ASoC: nuc900: Convert to devm_snd_soc_register_platform
ASoC: blackfin: Convert to devm_snd_soc_register_platform
ASoC: au1x: Convert to devm_snd_soc_register_platform
ASoC: qcom: Constify asoc_qcom_lpass_cpu_dai_ops
...
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell laptop has a series model to use the same codec but different subsystem ID.
At the same time they happens the white noise by login screen and headphone;
for fixing them together, I only can add these IDs to FIXUP function ALC292_FIXUP_DISABLE_AAMIX,
then try to solve such the similar issues.
Codec: Realtek ALC3235
Vendor Id: 0x10ec0293
Subsystem Id: 0x102806dd
Subsystem Id: 0x102806df
Subsystem Id: 0x102806e0
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1492132
Signed-off-by: Woodrow Shen <woodrow.shen@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The port_mst_index parameter was reserved for future use, but
maintainers prefer to add it later when it is actually used.
[Note: this is an update patch to commit [51e1d83cab99: drm/i915: Call
audio pin/ELD notify function] where I mistakenly applied the older
version. Jani and Daniel's review tags were to the latest version,
so I add them below, too -- tiwai]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Reviewed-by: Jani Nikula <jani.nikula@intel.com>
Acked-by: Daniel Vetter <daniel@ffwll.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Whenever there is an event from the i915 driver, wake the codec
and recheck plug/unplug + ELD status.
This fixes the issue with lost unsol events in power save mode,
the codec and controller can now sleep in D3 and still know when
the HDMI monitor has been connected.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This lets the interested codec be notified when an i915 pin/ELD
event happens.
[tiwai: Fixed a trivial build error for CONFIG_SND_HDA_I915=n]
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We were aborting if the kzalloc of img_swap fails but without freeing the
already allocated out. Similarly we were aborting if spi_sync fails
without releasing out and img_swap.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In case of unknown DT compatible device the ASRC OF node
possibly acquired earlier by of_parse_phandle() has
to be put before returning from probe method.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
Not many updates to the core here, but an awful lot of driver updates
this time round:
- Factoring out of AC'97 reset code into the core
- New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
RT298 and ST STI controllers.
- Machine drivers for Rockchip systems with MAX98090 and RT5645 and
RT5650.
- Initial driver support for Intel Skylake devices.
- A large number of cleanups for Lars-Peter Clausen and Axel Lin.
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Merge tag 'asoc-v4.2-rc8' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v4.3
Not many updates to the core here, but an awful lot of driver updates
this time round:
- Factoring out of AC'97 reset code into the core
- New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek
RT298 and ST STI controllers.
- Machine drivers for Rockchip systems with MAX98090 and RT5645 and
RT5650.
- Initial driver support for Intel Skylake devices.
- A large number of cleanups for Lars-Peter Clausen and Axel Lin.
Use devm_* API to fix leaks in current code.
1. Use devm_kzalloc to fix memory leak for zx_i2s when unload the module.
2. Use devm_snd_soc_register_component to ensure component is unregistered
when unload the module.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Reviewed-by: Jun Nie <jun.nie@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use devm_snd_soc_register_component to ensure component is unregistered
when unload the module.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use devm_ioremap_resource() instead of open code.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
All the callers assume devm_spear_pcm_platform_register is a devm_ API, so
use devm_snd_dmaengine_pcm_register in devm_spear_pcm_platform_register.
Fixes: e1771bcf99 ("ASoC: SPEAr: remove custom DMA alloc compat function")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In PCM core, when hw_params() in each driver returns error, the state of
PCM substream is kept as 'open'. In this case, current drivers for sound
units on IEEE 1394 bus doesn't decrement substream counter in hw_free()
correctly. This causes these drivers to keep streams even if not
required.
This commit fixes this bug. When snd_pcm_lib_alloc_vmalloc_buffer()
fails, hw_params function in each driver returns without incrementing the
counter.
Reported-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A number of functions and structures in the sound subsystem had incomplete
and/or obsolete DocBook comments, leading to warnings when the docs were
built. Correct those comments so that we can enjoy our audio in the
absence of warning noise.
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
asoc_qcom_lpass_cpu_dai_ops is exported and used by multiple drivers,
make it const to prevent modifying it at run time.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/ics43432.c:66:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
CC: Ricard Wanderlof <ricard.wanderlof@axis.com>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The check of cval->cached should be zero-based (including master channel).
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These platform drivers have a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add i2c shutdown function to prevent the pop sound of the headphone while
the system is rebooting or shutdowning. It de-initials the jack detection
function, and it cannot be turned off in _BIAS_OFF. If we don't de-initial
it, the pop sound will be heard in the situation of powering off. And
replace the related register settings from magic number to meaningful
defined name.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add support for the InvenSense ICS-43432 I2S MEMS microphone.
This is a non-software-configurable MEMS microphone with I2S output.
Tested on a setup with a single ICS-43432 (the device itself supports
stereo operation using a hardware pin controlling left vs. right channel
output).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set buffer bytes step constraint to 128. A matching constraint has
already been set to period size. This helps PCM setup to tolerate ALSA
clients that set the PCM hw params in unusual order.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In theory, the device may get suspended even at runtime PM suspend.
Currently we don't save the mixer state for autopm, and it may bring
inconsistency.
This patch removes the special handling for autosuspend.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent fix of runtime PM for USB-audio driver, we got a
lockdep warning like:
=============================================
[ INFO: possible recursive locking detected ]
4.2.0-rc8+ #61 Not tainted
---------------------------------------------
pulseaudio/980 is trying to acquire lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
but task is already holding lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
This comes from snd_usb_autoresume() invoking down_read() and it's
used in a nested way. Although it's basically safe, per se (as these
are read locks), it's better to reduce such spurious warnings.
The read lock is needed to guarantee the execution of "shutdown"
(cleanup at disconnection) task after all concurrent tasks are
finished. This can be implemented in another better way.
Also, the current check of chip->in_pm isn't good enough for
protecting the racy execution of multiple auto-resumes.
This patch rewrites the logic of snd_usb_autoresume() & co; namely,
- The recursive call of autopm is avoided by the new refcount,
chip->active. The chip->in_pm flag is removed accordingly.
- Instead of rwsem, another refcount, chip->usage_count, is introduced
for tracking the period to delay the shutdown procedure. At
the last clear of this refcount, wake_up() to the shutdown waiter is
called.
- The shutdown flag is replaced with shutdown atomic count; this is
for reducing the lock.
- Two new helpers are introduced to simplify the management of these
refcounts; snd_usb_lock_shutdown() increases the usage_count, checks
the shutdown state, and does autoresume. snd_usb_unlock_shutdown()
does the opposite. Most of mixer and other codes just need this,
and simply returns an error if it receives an error from lock.
Fixes: 9003ebb13f ('ALSA: usb-audio: Fix runtime PM unbalance')
Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The variable pcm can be never NULL since it was rewritten with
list_for_each_entry().
Suggested-by: Markus Osterhoff <linux-kernel@k-raum.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/soc/rockchip/rockchip_rt5645.c:214:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/rockchip/rockchip_max98090.c:225:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Being careless, judge the return value of snd_soc_card_jack_new
is opposite, so it should be fixed.
Signed-off-by: Xing Zheng <zhengxing@rock-chips.com>
Reviewed-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_hdac_refresh_widget_sysfs() is called before the first
hda_widget_sysfs_init(), the next call overrides and eventually
fails. This results in unexpected Oops, something like:
BUG: unable to handle kernel NULL pointer dereference at 00000000000000c8
IP: [<ffffffff8180e2a3>] hdmi_chmap_ctl_info+0x23/0x40
The fix is to add a check of the existing sysfs tree. Also, for more
safety, this patch adds the checks of device_is_registered() in
snd-hdac_refresh_wdiget_sysfs(), too.
Fixes: fa4f18b4f4 ('ALSA: hda - Refresh widgets sysfs at probing Haswell+ HDMI codecs')
Bugizlla: https://bugzilla.kernel.org/show_bug.cgi?id=103431
Reported-by: Andreas Reis <andreas.reis@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The extcon driver takes the DAPM mutex from within the interrupt thread
in several places, which makes it possible to get into a situation where
the interrupt thread is blocked waiting on the DAPM mutex whilst a DAPM
sequence is running which is attempting to configure the FLL. In this
case the FLL completion can't be completed as as the IRQ handler is
ONE_SHOT, which cause the FLL lock to use the full time out (250mS) and
report that the process timed out.
It is not really practical to make the extcon driver not take the DAPM
mutex from within the interrupt thread, at least not without extensive
modification. So this patch fixes the issue by switching the wait for
the FLL lock to polling. A few fast polls are done first as the FLL
should lock quickly for a good quality reference clock, (indeed it hits
on the first poll on my system) and it will poll every 20mS after that
until it times out.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The patch removes the incorrect setting of the JD mode. It will cause pop
sound in the booting time.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch corrects the sequence of the jack detection. It will prevent the
pop sound while the jack plug in.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Remove the original calibration function and modify the depop and
calibration function to prevent the pop sound in the booting time.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added rt5677_spi_read() and refactored rt5677_spi_write() so that
an arbitrary block in the DSP address space can be read/written
via SPI. For example, this allows us to load an ELF DSP firmware
with sparse sections, and stream audio samples from DSP ring buffer.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Acked-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Otherwise it triggers a compile warning like:
sound/ppc/keywest.c:104:1: warning: data definition has no type or storage class
sound/ppc/keywest.c:104:1: error: type defaults to 'int' in declaration of 'MODULE_DEVICE_TABLE' [-Werror=implicit-int]
Fixes: a2bc2af66a ('ALSA: ppc: keywest: Export I2C module alias information')
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Reviewed-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The I2C core always reports the MODALIAS uevent as "i2c:<client name"
regardless if the driver was matched using the I2C id_table or the
of_match_table. So the driver needs to export the I2C table and this
be built into the module or udev won't have the necessary information
to auto load the correct module when the device is added.
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix below build warning:
CC [M] sound/soc/tegra/tegra20_spdif.o
sound/soc/tegra/tegra20_spdif.c: In function 'tegra20_spdif_platform_remove':
sound/soc/tegra/tegra20_spdif.c:361:24: warning: unused variable 'spdif' [-Wunused-variable]
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Reviewed-by: Vaishali Thakkar <vthakkar1994@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use devm_ioremap_resource() instead of open code.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use devm_ioremap_resource() instead of open code.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/cs4349.c:389:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
CC: Tim Howe <tim.howe@cirrus.com>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the commit [fa4f18b4f402: ALSA: hda - Refresh widgets sysfs at
probing Haswell+ HDMI codecs], snd_hdac_refresh_widget_sysfs() is
explicitly called in the codec driver. But this results in refreshing
twice, as snd_hdac_refresh_widget_sysfs() itself calls
snd_hdac_refresh_widgets() function.
Instead, we can replace the call in snd_hda_codec_update_widgets()
with snd_hdac_refresh_widget_sysfs(). This also fixes the missing
sysfs update for ca0132, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The widget power-saving code tries to turn up/down the power of each
widget in the I/O paths that are modified at each jack plug/unplug.
The recent report revealed that the power activation leaves some
widgets unpowered after plugging. This is because
snd_hda_activate_path() turns on path->active flag at the end of the
function while the path power management is done before that. Then
it's regarded as if nothing is active, and the driver turns off the
power.
The fix is simply to set the flag at the beginning of the function,
before trying to power up.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102521
Cc: <stable@vger.kernel.org> [v4.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The is_active_nid_for_any() function in the generic parser is supposed
to check all connections from/to the given widget, but the current
code checks only the first input connection (index = 0).
This patch corrects the code to check all inputs by passing -1 to
index argument.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102521
Cc: <stable@vger.kernel.org> [v4.1+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After a for-loop was replaced by list_for_each_entry, see
Commit bbbc7e8502 ("ALSA: hda - Allocate hda_pcm objects dynamically"),
Commit 751e221689 ("ALSA: hda: fix possible null dereference"),
a possible NULL pointer dereference has been introduced; this patch adds
the NULL check on pcm->pcm, while leaving a potentially superfluous
check on pcm itself untouched.
Signed-off-by: Markus Osterhoff <linux-kernel@k-raum.org>
Cc: <stable@vger.kernel.org> #v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a helper to find the stream using stream tag and direction.
This is useful for drivers to query stream based on stream tag
and direction, fox example while downloading FW thru DSP loader
code
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
spbmaxfifo API is actually a query function not a set function so
name it snd_hdac_ext_stream_get_spbmaxfifo()
Reported-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While creating hdac_ext_device, we used hdev for sizeof insteadof
edev, which resulted in eventual crash of the system Fix the size
here
Fixes: a512f56116 ('ALSA: hdac: add hdac extended device')
Reported-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
card->shortname is a 32 char string so the sprintf() can theoretically
overflow. snd_rawmidi_new() can accept strings up to 64 bytes long.
I have made the temporay buf[] array 40 bytes long and changed the
sprintf() to snprintf().
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow vendor drivers to define bespoke bytes ext handlers and IDs for
TLV bytes controls. And the topology core will bind these handlers by
matching IDs defined by the vendor driver and user space topology
data file.
And TLV callback binding is moved to soc_tplg_kcontrol_bind_io(). This
function process all handler binding now.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the pointer of struct soc_tplg as one argument, so no need to
pass standard/vendor specific kcontrol handlers and their count.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Vendor specific handlers should override standard handlers. So we can
handle things in the order from specific to generic.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
hdac_ext_stream assign doesn't require key mapping as in case of
hdac_stream. So for host stream, the key to device mapping needs
to be removed.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The drivers need to set the spib and maxfifios values, so add
these new APIs snd_hdac_ext_stream_set_spib() and
snd_hdac_ext_stream_set_spbmaxfifo() APIs
For these APIs we also need to have spib and fifos pointer, so
add these to hdac_ext_stream and initialize them at stream init
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
New HDA controllers like Skylake sport multiple HDA links, so we need a
helper to turn off all the links in one go while suspending the device so
add snd_hdac_ext_bus_link_power_down_all() API
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
SPCAP and Mutilink register offset were incorrect as offset needs
to be based on capability offset. So correct the offset for
read/write of spcap/link register.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the ac97_bus from legacy suspend/resume callbacks to dev_pm_ops.
Since there isn't anything special to do at the bus level the bus driver
does not have to implement any callbacks. The device driver core will
automatically pick up and execute the device's PM ops.
As there is only a single AC'97 driver implementing suspend and resume,
update both the core and driver at the same time to avoid unnecessary code
churn.
While we are at it also drop the ifdefs around the suspend/resume functions
to increase compile test coverage.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Dmitry Torokhov <dmitry.torokhov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set the card owner field to prevent the module from being removed from
underneath its users.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel Haswell (and later) codec refreshes the widgets tree to expose
the whole widget nodes at probing. However, this refresh was missing
for sysfs tree.
This patch adds the missing piece, as we have now a proper API.
Reported-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I believe this probably cannot happen, as the code suggests. There
would have to be an kcontrol->index.id which was zero, otherwise this
would be prevented in snd_ctl_find_id(). But snd_BUG_ON() is just a
WARN() or a no-op so static checkers complain that we keep on going with
a negative offset. Let's just handle the error as well as printing
a warning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some codecs like Intel HDMI by default do not show up all the pins, they
have to be manually enabled, so we need to refresh the codec widgets and
then recreate the sysfs tree. So add new API snd_hdac_refresh_widget_sysfs()
to do this. It should be be used by codec driver after sending magic verbs
to codec
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds new extended driver objects and API for registering the
extended devices.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds based hdac extended device object which will be used by
ASoC HDAC codecs
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HDAC extended device objects are created by HDAC extended bus on probe.
When controller is removed they should be removed as well, so add API
snd_hdac_ext_bus_device_remove for this
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On shutdown/reboot of CX20722, first shut down all EAPDs, then
shut down the afg node to D3.
Failure to do so can lead to spurious noises from the internal speaker
directly after reboot (and before the codec is reinitialized again, i e
in BIOS setup or GRUB menus).
BugLink: https://bugs.launchpad.net/bugs/1487345
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Gustard DAC-X20U.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If there don't exist dynamic sink or source widget, it will failed to
add dynamic path.
"AIF3ADCDAT" is snd_soc_dapm_aif_out, can't be dynamic sink widget. So
change the audio route to fix this issue.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no use of snd_soc_unregister_card in remove function
as devm_snd_soc_register_card in probe function automatically
handles it. So, remove use of snd_soc_unregister_card and with
this change remove arndale_audio_remove as it is now redundant.
Signed-off-by: Vaishali Thakkar <vthakkar1994@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The topology code merged in the v4.2 merge window introduced a new ABI
which was believed to be suitable for use but subsequently additional
work by the developers of this feature have revealed some problems that
need to be addressed. In order to allow this to be done without having
to support the initial ABI add Kconfig to disable the build and also add
some #error statements to the UAPI header so users can't use them.
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Merge tag 'asoc-v4.2-disable-topology' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Disable topology support for v4.2
The topology code merged in the v4.2 merge window introduced a new ABI
which was believed to be suitable for use but subsequently additional
work by the developers of this feature have revealed some problems that
need to be addressed. In order to allow this to be done without having
to support the initial ABI add Kconfig to disable the build and also add
some #error statements to the UAPI header so users can't use them.
The wrong register was used to set the gain of ref loop, when changing
the FLL output on an active FLL. This patch corrects the offset of the
gain register.
Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The input terminal parser recurses into the referenced clock entity to verify
it is existant and thus the terminal descriptor is valid. The actual property
values of the term instance which is initially parsed must not be overriden by
the recursion. For this to work the term properties have to be assigned after
recursing into the referenced clock entity descriptors.
Signed-off-by: Julian Scheel <julian@jusst.de>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove use of snd_soc_unregister_component in remove function
as devm_snd_soc_register_component in probe function automatically
handles it.
Also, convert call of snd_dmaengine_pcm_register to managed resource
function devm_snd_dmaengine_pcm_register and remove usage of
snd_dmaengine_pcm_unregister in probe and remove functions.
Signed-off-by: Vaishali Thakkar <vthakkar1994@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The fix for deadlock in PM in commit [1ee23fe07ee8: ALSA: usb-audio:
Fix deadlocks at resuming] introduced a new check of in_pm flag.
However, the brainless patch author evaluated it in a wrong way
(logical AND instead of logical OR), thus usb_autopm_get_interface()
is wrongly called at probing, leading to unbalance of runtime PM
refcount.
This patch fixes it by correcting the logic.
Reported-by: Hans Yang <hansy@nvidia.com>
Fixes: 1ee23fe07e ('ALSA: usb-audio: Fix deadlocks at resuming')
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Revert the problematic part of commit 470805eb9f ("ASoC: tegra:
Convert to managed resources"). Before this commit, PM cleanup was
performed after the component was unregistered. But returning
directly will skip PM cleanup. So, to be on safe side it is better
to use snd_soc_register_component instead of
devm_snd_soc_register_component.
Signed-off-by: Vaishali Thakkar <vthakkar1994@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
regmap_readable() returns false if map->format.format_write is set.
For .reg_bits = 7, .val_bits = 9, setting,
map->format.format_write = regmap_format_7_9_write;
Even current code has implemented map->readable_reg, regmap_readable()
still returns false anyway. Thus drop the misleading readable_reg callback
implementation.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Allow the topology code to be compiled out so that users who don't need
topology don't need to havve the code compiled in, saving them some
memory.
Some more configuration could be added to remove some of the hooks into
the core data structures but that is probably best done with some
refactoring to use functions to do the updates of the data structures
rather than ifdefing in the code as we'd need to do at the minute.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use resource managed function devm_snd_soc_register_component for
component registration instead of snd_soc_register_component.
Also, remove davinci_vcif_remove as it is now redundant.
Signed-off-by: Vaishali Thakkar <vthakkar1994@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
DAPM core already creates widgets for DAIs. It is not necessary
to declare them by SND_SOC_DAPM_AIF_IN/SND_SOC_DAPM_AIF_OUT.
Furthermore, original codes use backend DAI's stream name to be the AIF
widget name. It causes the same widget to be created twice, and after
commit 92fa124267 ("ASoC: dapm: Add new widgets to the end of the
widget list") the first created widget (by snd_soc_dapm_new_controls)
is used, not the 2nd created one (by snd_soc_dapm_new_dai_widgets),
so audio path is broken.
Signed-off-by: Koro Chen <koro.chen@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The power states in a proc file are printed in a racy manner on a
single static string buffer. Fix it by calling snd_iprintf() directly
for each state instead of processing on a temporary buffer.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Many arrays in hda_proc.c are string arrays that should be covered by
const prefix for increasing the safety and reducing the size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few helper functions to convert the pin information to strings have
been exported with assumption that they were used by other drivers.
But they are referred only in the proc interface in the end.
Let's make them local so that we can get rid of a few exports.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M-Audio Transit exposes an interface with a SYNC_NONE attribute.
This is not a valid value according to the USB audio classspec. However
there is a sync endpoint associated to this record. Changing the logic to
try to use this sync endpoint allows for seamless transitions between
altset 2 and altset 3. If any errors happen, the behavior remains the same.
$ more /proc/asound/card1/stream0
M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (ADAPTIVE)
Rates: 48001 - 96000 (continuous)
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (NONE)
Rates: 8000 - 48000 (continuous)
Interface 1
Altset 3
Format: S16_LE
Channels: 2
Endpoint: 3 OUT (ASYNC)
Rates: 8000 - 48000 (continuous)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a transition occurs between alternate settings that do not use the
same synchronization method, the substream pointers were not reset.
This prevents audio from being played during the second transition.
Identified and tested with M-Audio Transit device
(0763:2006 Midiman M-Audio Transit)
Details of the issue:
First playback to adaptive endpoint:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
[ 3169.297556] usb 1-2: setting usb interface 1:1
[ 3169.297568] usb 1-2: Creating new playback data endpoint #3
[ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0
[ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000
first playback to asynchronous endpoint:
$ aplay -Dhw:1,0 ~/16_48.wav
Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian,
Rate 48000 Hz, Stereo
[ 3204.520251] usb 1-2: setting usb interface 1:3
[ 3204.520264] usb 1-2: Creating new playback data endpoint #3
[ 3204.520272] usb 1-2: Creating new capture sync endpoint #83
[ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
[ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000
[ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000
second playback to adaptive endpoint: no audio and error on terminal:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
aplay: pcm_write:1939: write error: Input/output error
[ 3239.483589] usb 1-2: setting usb interface 1:1
[ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000
[ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
This last line shows that a sync endpoint is used when it shouldn't.
The sync endpoint is no longer valid and the pointers are corrupted
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
add wm8960 support for fsl-asoc-card
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add OF match table to SSM2518 to allow direct matching without going
through I2C subsystem.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The PCM DAIs need to be loaded and added to ASoC core ealier than the
graph (route). Otherwise, adding routes will fail for missing DAIs.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adjust set DAI format function in fsl_ssi driver
so it doesn't fail and clears RXDIR in AC'97 mode.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instantiate AC'97 CODEC in fsl_ssi driver AC'97 mode.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
Check whether setting AC'97 ops succeeded and clean them
on removal so the fsl_ssi driver can be reloaded.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
AC'97 bus can support asymmetric playback/capture rates
so enable them in this case in fsl_ssi driver.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
AC'97 DAI driver struct need the same probe method as
I2S one to setup DMA params in fsl_ssi driver.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
IPG clock have to be enabled during AC'97 CODEC register
access in fsl_ssi driver.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
Signed-off-by: Mark Brown <broonie@kernel.org>
pm_runtime_get_sync() increments the runtime PM usage counter even the
call returns an error code. Thus a pairing decrement is needed on the
error handling path to keep the counter balanced.
Signed-off-by: Junjie Mao <junjie.mao@enight.me>
Signed-off-by: Mark Brown <broonie@kernel.org>
The readable registers are in continuous ranges: 0x01 ~ 0x03, 0x05 ~ 0x5f.
Use case range syntax makes the code shorter with better readability when
we have a large number of continuous switch cases.
No functional change with this patch.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The readable registers are in continuous range: 0x01 ~ 0x2e.
Use case range syntax makes the code shorter with better readability when
we have a large number of continuous switch cases.
No functional change with this patch.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The readable registers are in continuous range: 0x01 ~ 0x34.
Use case range syntax makes the code shorter with better readability when
we have a large number of continuous switch cases.
No functional change with this patch.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use case range syntax makes the code shorter with better readability when
we have a large number of continuous switch cases.
Below are the summary of readable/volatile/precious registers.
The readable registers:
0x01 ~ 0x0D, 0x0F ~ 0x1C
The volatile registers:
0x01 ~ 0x05, 0x15 ~ 0x18
The precious registers:
0x15 ~ 0x18
No functional change with this patch.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use managed resource functions devm_clk_put and
devm_snd_soc_register_component to simplify error handling.
To be compatible with the change various gotos are replaced
with direct returns, and unneeded labels are dropped.
Signed-off-by: Vaishali Thakkar <vthakkar1994@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ssm4567 has sensing circuitry that can be used to monitor the current
and voltage on the speaker amplifier output has well as the VBAT input.
This data can be output over the I2S interface so it can be processed by a
DSP or similar.
This patch adds the sense capture output stream to the CODEC DAI as well as
DAPM widgets that ensure that the sensing circuitry is powered up when the
capture stream is active.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Don't set .read_flag_mask for adav803, it's for adav801 only.
Fixes: 0c2d696456 ("ASoC: adav80x: Split SPI and I2C code into different modules")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
USB Audio Class version 2.0 supports three different parameter block sizes for
CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes
(5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter
Block). Use the correct size according to the specific control as it was
already done for UACv1. The allocated block size for control requests is
increased to support the 4 byte worst case.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The dev->name of CODEC might not be identical to its codec_dai_name,
so using dev->name to probe the CODEC dai is not a correct approach.
This patch specifies each supporting codec_dai_name instead of using
dev->name any more.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The snd_soc_dapm_input_path and snd_soc_dapm_output_path trace events are
identical except for the direction. Instead of having two events have a
single one that has a field that contains the direction.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
After the recent cleanups and generalizations of the DAPM algorithm the
handling of input and output paths is now fully symmetric. This means by
making some slight changes to the data structure and using arrays with one
entry for each direction, rather than separate fields, it is possible to
create a generic implementation that is capable of handling both input and
output paths.
Unfortunately this generalization significantly increases the code size on
the hot path of is_connected_{input,output}_ep() and
dapm_widget_invalidate_{input,output}_paths(), which has a negative impact
on the overall performance. The inner loops of those functions are quite
small and the generic implementation adds extra pointer arithmetic in a few
places.
Testing on ARM shows that the combined code size of the specialized
functions is about 50% larger than the generalized function in relative
numbers. But in absolute numbers its less than 200 bytes, which is still
quite small. On the other hand the generalized function increases the
execution time of dapm_power_one_widget() by 30%. Given that this function
is one of the most often called functions of the DAPM framework the
trade-off of getting better performance at expense of generating slightly
larger code at seems to be worth it.
To avoid this still keep two versions of these functions around, one for
input and one for output. But have a generic implementation of the
algorithm which gets inlined by those two versions. And then let the
compiler take care of optimizing it and removing he extra instructions.
This still reduces the source code size as well as the makes making changes
to the implementation more straight forward since the same change does no
longer need to be done in two separate places. Also on the slow paths we
can use a generic implementations that handle both input and output paths.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make sure to unlock the DAPM mutex when dapm_widget_list_create() fails.
This means the function will now generate a trace_snd_soc_dapm_connected
event, even if the creation of the list fails. But that was the behavior
before the patch that introduced the unlock issue, so that should be fine.
Fixes: 1ce43acff0 ("ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets()")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Dell laptop causes the white noise by login screen and headphone,
and the fixup function ALC292_FIXUP_DISABLE_AAMIX can eliminate this
noise.
Codec: Realtek ALC3235
Vendor Id: 0x10ec0293
Subsystem Id: 0x102806db
Cc: <stable@vger.kernel.org>
BugLink: https://bugs.launchpad.net/bugs/1484334
Signed-off-by: Woodrow Shen <woodrow.shen@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are too much noise about the typos for fsl's drivers. So I fix
all the typos here in this patch in almost every file I touched.
Signed-off-by: Xiubo Li <lixiubo@cmss.chinamobile.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add 32 bit word length support. There are no code changes required
in the SAI driver since it has already wirten the word width to the
corresponding register.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The power for line out was not turned on when line out is enabled.
So we add "LOUT amp" widget to turn on the power for line out.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
On arm64:
sound/soc/sh/rcar/dma.c: In function 'rsnd_dmaen_init':
sound/soc/sh/rcar/dma.c:180:9: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast]
(void *)id);
^
include/linux/dmaengine.h:1185:75: note: in definition of macro 'dma_request_channel'
#define dma_request_channel(mask, x, y) __dma_request_channel(&(mask), x, y)
^
Add an intermediate cast to "uintptr_t" to kill the compile warning.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add sysclk auto mode. When it's sysclk auto mode, if the MCLK is
available for clock configure, using MCLK to provide sysclk directly,
otherwise, search a available pll out frequcncy and set pll.
Configure clock in hw_params may cause problems when using bypass style
paths without hw_params in machine driver getting called. So add configure
clock to set_bias_level.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Alienware 15 has CA0132 codec for its onboard sound, but the pin
config and mapping seem quite different from other Creative boards.
This patch corrects them, at least, for providing the right headphone
and mic jack notification, as well as removing the non-existing SPDIF
pins.
Even with this fix, not all stuff works perfectly yet, mainly because
of the badly written ca0132 driver code -- it has too many implicit
assumptions of pin configs and maps. Nevertheless, this is a small
good step forward.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101981
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The echoaudio locally defines TRUE and FALSE. Not only is this
redundant given that C now has a boolean type it results in lots of
warnings as other headers also define these macros, causing duplicate
definitions. Fix this by removing the local defines and converting all
local users to use the standard C true and false instead, simply
removing the macros is less safe due to implicit inclusion of the other
definitons.
[fixed overlooked replacement of FALSE by tiwai]
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enforce correct device sequencing when configuring a new
audio route when there is an existing active audio route(s).
This patch fixed recording noise issue while playback is active.
We have some registers which require the device to be in full shutdown
or to enter full shutdown before the register settings will take effect.
Currently the driver is not shutting down the device when a new audio
route is created. If a new audio route is made active while there is
already an active audio route, then the required register sequencing is
violated. A hardware shutdown toggle when creating a new audio route
corrects the sequencing error. The device must remain in hardware
shutdown for 40ms to allow the internal hardware core to fully shutdown.
Signed-off-by: Fang, Yang A <yang.a.fang@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Acked-by: Anish Kumar <anish.kumar@maximintegrated.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To manage DSP we need to create processing pipeline and on cleanup destroy
them. So we add create and destroy routines for pipelines The pipelines need
to to be executed so we add pipeline run and stop routines
All these send required IPCs to DSP using IPC routines added earlier
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A module needs to be instantiated and then connected with other modules. On
cleanup we need to disconnect the module.
This is achieved by helpers module init, bind and unbind which are added
here
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SRC and converter modules are required to do frequency and channel
conversion in DSP. Both take base module configuration and additional SRC
and converter parameters. The helpers here are added to calculate the values
for these modules
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds helper functions to calculate parameters required for base module
format and copier module. A generic module is modelled by base module.
Copier module is responsible for getting/sending data to FE (host DMAs) and
BE (link HDA DMA, SSP, PDM)
This also ads module pin management helpers which help in finding pins to
use or freeing them up
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cyrille Pitchen