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>
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>
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1
iQEcBAABAgAGBQJWBF7yAAoJECTWi3JdVIfQ1MEH/jnzSyEVIuG+l8UkMaz6gf4w
zGsM1KCn//mfPl7yAoOdsnElOLR+Fmf+0jx4pCPQKrjvBGwjwH/IwBR1rwuEeUPY
7d66efpWOKlTf3qpsF1S7ZIlAZOs0NFvo0jwA1ZY/pc3YEBekyWxbABk/uWAVrM5
HJJKafI7WeiYrF0l0z2sG7BpsFtr8JKqrOVM+SGaPTNn2k+/lQ1bwTk1liOEUbsv
oq8NFNrUWPBCwbUNJQxBOvmoXC6Oa6+JBVO3+SsoS0q2FweNpqtZopjmoqHM8CiN
SkBeFT+wYlSGSnnFgAXXA2+kq74TeP2CvToo6tw+gf4LZXydKIaAdeuT6M9weZA=
=8h3u
-----END PGP SIGNATURE-----
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.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1
iQEcBAABAgAGBQJWBEMrAAoJECTWi3JdVIfQOqIH/jsO0wdDz683ZpUd0K3OQlss
gia5/e0pS4IOaQY4ECZSydC/wf+fGs0ZHlLWXqSzJ33abCUUZlfL4f/3kQwhIrgD
Tb4aFLQoTRglZIqsgEm91Mqpk9gFUxhhqRBhI77iw11SOG1uWdokkYISG0ljnR5p
HFVxmqiSubvKdtydTOWR446Gxrk97c8HjzoBOXvQ87hKKyos7oJi4OcYD6HDVNr9
hrPkHS/05anaLbehZr82jmL+yMDsQl7QMjk1ljRkuufDUB07HogM1FHb5zkecC9u
eqDy5SOSJY4XFINDpxqt/5nqDaKgPcbEpfCH+ajfeY0e3d8rVVnPurrz/H4ElUM=
=KbEn
-----END PGP SIGNATURE-----
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 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>
