Current rsnd driver is using rsnd_kctrl_new_m/s/e function,
but the differences are very few.
This patch merge these rsnd_kctrl_new_m/s/e into rsnd_kctrl_new
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current src->convert_rate will be set on .hw_param, and
be reset on .quit timing.
But, .hw_param will not be called again if user did Ctrl-Z + fg.
It should be reset on initial of .hw_param to keep its value.
Here, ctu.c already do this.
This patch solves this issue, other wise, MIXed sound will be
strange if user did like below.
> aplay -D plughw:0,0 sound_44100.wav &
> aplay -D plughw:0,1 sound_96000.wav
> Ctrl-Z
> fg # 96kHz will be played as 44.1kHz
Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
In ALSA firewire stack, some AV/C commands are supported, including
vendor's extensions. Drivers includes response parser of each command,
according to its requirements, while the parser is written with loose
fashion in two points; error check and length check. This doesn't cause
any issues such as kernel corruption, but should be improved.
This commit modifies evaluations of return value on each parsers.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In protocol version 3, drivers can read current sampling clock status from
register 0x'ffff'f000'0b14. 8 bits of LSB of this register represents type
of signal as source of clock.
Current driver code includes invalid bitshift to handle the parameter. This
commit fixes the bug.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Fixes: 5992e30034c4 ("ALSA: firewire-motu: add support for MOTU 828mk3 (FireWire/Hybrid) as a model with protocol version 3")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At a commit 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound
card"), ALSA oxfw driver fails to handle SCS.1m/1d, due to -EBUSY at a call
of snd_card_register(). The cause is that the driver manages to register
two rawmidi instances with the same device number 0. This is a regression
introduced since kernel 4.7.
This commit fixes the regression, by fixing up device property after
discovering stream formats.
Fixes: 6c29230e2a5f ("ALSA: oxfw: delayed registration of sound card")
Cc: <stable@vger.kernel.org> # 4.7+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For digi00x series, asynchronous transaction is not used to transfer MIDI
messages to/from control surface. One of transction handlers in my previous
work loses its practical meaning.
This commit removes the handler. I note that unit of console type
transfers 0x00001000 to registered address of host space when switching
to 'standalone' mode. Then the unit generates bus reset.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At a commit c5fcee0373b3 ("ALSA: firewire-digi00x: add MIDI operations for
MIDI control port"), I described that MIDI messages for control surface is
transferred by a different way from the messages for physical ports.
However, this is wrong. MIDI messages to/from all of MIDI ports are
transferred by isochronous packets.
This commit removes codes to transfer MIDI messages via asynchronous
transaction, from MIDI handling layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At a commit 9dc5d31cdceb ("ALSA: firewire-digi00x: handle MIDI messages in
isochronous packets"), a functionality to handle MIDI messages on
isochronous packet was supported. But this includes some of my
misunderstanding. This commit is to fix them.
For digi00x series, first data channel of data blocks in rx/tx packet
includes MIDI messages. The data channel has 0x80 in 8 bit of its MSB,
however it's against IEC 61883-6. Unique data format is applied:
- Upper 4 bits of LSB represent port number.
- 0x0: port 1.
- 0x2: port 2.
- 0xe: console port.
- Lower 4 bits of LSB represent the number of included MIDI message bytes;
0x0/0x1/0x2.
- Two bytes of middle of this data channel have MIDI bytes.
Especially, MIDI messages from/to console surface are also transferred by
isochronous packets, as well as physical MIDI ports.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi00x series includes two types of unit; rack and console. As long as
reading information on config rom of Digi 002 console, 'MODEL_ID' field
has a different value from the one on Digi 002 rack.
We've already got a test report from users with Digi 003 rack. We can
assume that console type and rack type has different value in the field.
This commit adds a device entry for console type. For following commits,
this commit also adds a member to 'struct snd_digi00x' to identify console
type.
$ cd linux-firewire-utils/src
$ python2 ./crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 0404f9d0 bus_info_length 4, crc_length 4, crc 63952
404 31333934 bus_name "1394"
408 60647002 irmc 0, cmc 1, isc 1, bmc 0, cyc_clk_acc 100, max_rec 7 (256)
40c 00a07e00 company_id 00a07e |
410 00a30000 device_id 0000a30000 | EUI-64 00a07e0000a30000
root directory
-----------------------------------------------------------------
414 00058a39 directory_length 5, crc 35385
418 0c0043a0 node capabilities
41c 04000001 hardware version
420 0300a07e vendor
424 81000007 --> descriptor leaf at 440
428 d1000001 --> unit directory at 42c
unit directory at 42c
-----------------------------------------------------------------
42c 00046674 directory_length 4, crc 26228
430 120000a3 specifier id
434 13000001 version
438 17000001 model
43c 81000007 --> descriptor leaf at 458
descriptor leaf at 440
-----------------------------------------------------------------
440 00055913 leaf_length 5, crc 22803
444 000050f2 descriptor_type 00, specifier_ID 50f2
448 80000000
44c 44696769
450 64657369
454 676e0000
descriptor leaf at 458
-----------------------------------------------------------------
458 0004a6fd leaf_length 4, crc 42749
45c 00000000 textual descriptor
460 00000000 minimal ASCII
464 44696769 "Digi"
468 20303032 " 002"
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fireface 400 is a second model of RME Fireface series, released in 2006.
This commit adds support for this model.
This model supports 8 analog channels, 2 S/PDIF channels and 8 ADAT
channels in both of tx/rx packet. The number of ADAT channels differs
depending on each mode of sampling transmission frequency.
$ python2 linux-firewire-utils/src/crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 04107768 bus_info_length 4, crc_length 16, crc 30568 (should be 61311)
404 31333934 bus_name "1394"
408 20009002 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 0, max_rec 9 (1024)
40c 000a3501 company_id 000a35 |
410 1bd0862a device_id 011bd0862a | EUI-64 000a35011bd0862a
root directory
-----------------------------------------------------------------
414 000485ec directory_length 4, crc 34284
418 03000a35 vendor
41c 0c0083c0 node capabilities per IEEE 1394
420 8d000006 --> eui-64 leaf at 438
424 d1000001 --> unit directory at 428
unit directory at 428
-----------------------------------------------------------------
428 000314c4 directory_length 3, crc 5316
42c 12000a35 specifier id
430 13000002 version
434 17101800 model
eui-64 leaf at 438
-----------------------------------------------------------------
438 000261a8 leaf_length 2, crc 25000
43c 000a3501 company_id 000a35 |
440 1bd0862a device_id 011bd0862a | EUI-64 000a35011bd0862a
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds hwdep interface so as the other drivers for audio and
music units on IEEE 1394 have.
This interface is designed for mixer/control applications. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds PCM functionality to transmit/receive PCM frames on
isochronous packet streaming. This commit enables userspace applications
to start/stop packet streaming via ALSA PCM interface.
Sampling rate requested by applications is used as sampling transmission
frequency of IEC 61883-1/6packet streaming. As I described in followed
commits, units in this series manages sampling clock frequency
independently of sampling transmission frequency, and they supports
resampling between their packet streaming/data block processing layer and
sampling data processing layer. This commit take this driver to utilize
these features for usability.
When internal clock is selected as source signal of sampling clock, this
driver allows user space applications to start PCM substreams at any rate
which packet streaming engine supports as sampling transmission frequency.
In this case, this driver expects units to perform resampling PCM frames
for rx/tx packets when sampling clock frequency and sampling transmission
frequency are mismatched. This is for daily use cases.
When any external clock is selected as the source signal, this driver
gets configured sampling rate from units, then restricts available
sampling rate to the rate for PCM applications. This is for studio use
cases.
Models in this series supports 64.0/128.0 kHz of sampling rate, however
these frequencies are not supported by IEC 61883-6 as sampling transmission
frequency. Therefore, packet streaming engine of ALSA firewire stack can't
handle them. When units are configured to use any external clock as source
signal of sampling clock and one of these unsupported rate is configured
as rate of the sampling clock, this driver returns EIO to user space
applications.
Anyway, this driver doesn't voluntarily configure parameters of sampling
clock. It's better for users to work with appropriate user space
implementations to configure the parameters in advance of usage.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds management functionality for packet streaming.
As long as investigating Fireface 400, there're three modes depending
on sampling transmission frequency. The number of data channels in each
data block is different depending on the mode. The set of available
data channels for each mode might be different for each protocol and
model.
The length of registers for the number of isochronous channel is just
three bits, therefore 0-7ch are available.
When bus reset occurs on IEEE 1394 bus, the device discontinues to
transmit packets. This commit aborts PCM substreams at bus reset handler.
As I described in followed commits, The device manages its sampling clock
independently of sampling transmission frequency against IEC 61883-6.
Thus, it's a lower cost to change the sampling transmission frequency,
while data fetch between streaming layer and DSP require larger buffer
for resampling. As a result, device latency might tend to be larger than
ASICs for IEC 61883-1/6 such as DM1000/DM1100/DM1500 (BeBoB),
DiceII/TCD2210/TCD2220/TCD3070 and OXFW970/971.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating Fireface 400, format of payload of each
isochronous packet is not IEC 61883-1/6, thus its format of data block
is not AM824. The remarkable points of the format are:
* The payload just consists of some data channels of quadlet size without
CIP header.
* Each data channels includes data aligned to little endian order.
* One data channel consists of two parts; 8 bit ancillary field and 24 bit
PCM frame.
Due to lack of CIP headers, rx/tx packets include no CIP headers and
different way to check packet discontinuity. For tx packet, the ancillary
field is used for counter. However, the way of counting is different
depending on positions of data channels. At 44.1 kHz, ancillary field in:
* 1st/6th/9th/10th/14th/17th data channels: not used for this purpose.
* 2nd/18th data channels: incremented every data block (0x00-0xff).
* 3rd/4th/5th/11th/12th/13th data channels: incremented every 256 data
blocks (0x00-0x07).
* 7th/8th/15th/16th data channels: incremented per the number of data
blocks in a packet. The increment can occur per packet (0x00-0xff).
For tx packet, tag of each isochronous packet is used for this purpose.
The value of tag cyclically changes between 0, 1, 2 and 3 in this order.
The interval is different depending on sampling transmission frequency.
At 44.1/48.0 kHz, it's 256 data blocks. At 88.2 kHz, it's 96 data blocks.
The number of data blocks in tx packet is exactly the same as
SYT_INTERVAL. There's no empty packet or no-data packet, thus the
throughput is not 8,000 packets per sec. On the other hand, the one in
rx packet is 8,000 packets per sec, thus the number of data blocks is
different between each packet, depending on sampling transmission
frequency:
* 44.1 kHz: 5 or 6
* 48.0 kHz: 5 or 6 or 7
* 88.2 kHz: 10 or 11 or 12
This commit adds data processing layer to satisfy the above specification
in a policy of 'best effort'. Although PCM frames are handled for
intermediate buffer to user space, the ancillary data is not handled at all
to reduce CPU usage, thus counter is not checked. 0 is always used for tag
of isochronous packet. Furthermore, the packet streaming layer is
responsible for calculation of the number of data blocks for each packet,
thus it's not exactly the same sequence from the above observation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating Fireface 400, IEC 61883-1/6 is not applied to
its packet streaming protocol. Remarks of the specific protocol are:
* Each packet doesn't include CIP headers.
* 64,0 and 128,0 kHz are supported.
* The device doesn't necessarily transmit 8,000 packets per second.
* 0, 1, 2, 3 are used as tag for rx isochronous packets, however 0 is
used for tx isochronous packets.
On the other hand, there's a common feature. The number of data blocks
transferred in a second is the same as sampling transmission frequency.
Current ALSA IEC 61883-1/6 engine already has a method to calculate it and
this driver can utilize it for rx packets, as well as tx packets.
This commit adds support for the transferring protocol. CIP_NO_HEADERS
flag is newly added. When this flag is set:
* Both of 0 (without CIP header) and 1 (with CIP header) are used as tag
to handle incoming isochronous packet.
* 0 (without CIP header) is used as tag to transfer outgoing isochronous
packet.
* Skip CIP header evaluation.
* Use unique way to calculate the quadlets of isochronous packet payload.
In ALSA PCM interface, 128.0 kHz is not supported, and the ALSA
IEC 61883-1/6 engine doesn't support 64.0 kHz. These modes are dropped.
The sequence of rx packet has a remarkable quirk about tag. This will be
described in later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Audio and music units of RME Fireface series use its own protocol for
isochronous packets to transfer data. This protocol requires ALSA IEC
61883-1/6 engine to have alternative functions.
This commit is a preparation for the protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drivers can retrieve the state and configuration of clock by read
transactions.
This commit allows protocol abstraction layer to to dump the
information for debugging, via proc interface.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, fireface driver supports unique transaction mechanism
for MIDI feature. This commit adds MIDI functionality for userspace
applications.
As I wrote in a followed commit, user space applications get some
requirement from this driver. It should not touch a register to which
units transmit MIDI messages. It should configure a register in which
MIDI transmission is controlled.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As long as investigating Fireface 400, MIDI messages are transferred by
asynchronous communication over IEEE 1394 bus.
Fireface 400 receives MIDI messages by write transactions to two addresses;
0x'0000'0801'8000 and 0x'0000'0801'9000. Each of two seems to correspond to
MIDI port 1 and 2.
Fireface 400 transfers MIDI messages by write transactions to certain
addresses which configured by drivers. The drivers can decide upper 4 byte
of the addresses by write transactions to 0x'0000'0801'03f4. For the rest
part of the address, drivers can select from below options:
* 0x'0000'0000
* 0x'0000'0080
* 0x'0000'0100
* 0x'0000'0180
Selected options are represented in register 0x'0000'0801'051c as bit
flags. Due to this mechanism, drivers are restricted to use addresses on
'Memory space' of IEEE 1222, even if transactions to the address have
some side effects.
This commit adds transaction support for MIDI messaging, based on my
assumption that the similar mechanism is used on the other protocols. To
receive asynchronous transactions, the driver allocates a range of address
in 'Memory space'. I apply a strategy to use 0x'0000'0000 as lower 4 byte
of the address. When getting failure from Linux FireWire subsystem, this
driver retries to allocate addresses.
Unfortunately, read transaction to address 0x'0000'0801'051c returns zero
always, however write transactions have effects to the other features such
as status of sampling clock. For this reason, this commit delegates a task
to configure this register to user space applications. The applications
should set 3rd bit in LSB in little endian order.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As of 2016, RME discontinued its Fireface series, thus it's OK for us
to focus on released firmwares to drive known units.
As long as investigating Fireface 400 with Windows driver and comparing
the result to FFADO implementation, I can see these firmwares have
different register assignments. On the other hand, according to manuals
of each models, features relevant to packet streaming seem to be common,
because GUIs for these models have the same options. It's reasonable to
assume an abstraction layer of protocols to communicate to each models.
This commit adds the abstraction layer for the protocols. This layer
includes some functions to operate common features of models in this
series.
In IEC 61883-1/6, the sequence of packet can transfer timing information
to synchronize receivers to transmitters. Units of each node on IEEE 1394
bus can generate transmitter's timing clock by handling value of SYT field
in CIP header with high-precision clock. For audio and music units on
IEEE 1394 bus, this recovered clock is designed to used for sampling clock
to capture/generate PCM frames on DSP/ADC/DAC. (Actually, in this world,
there's no units to implement this specification as is, as long as I
know).
Fireface series doesn't use this mechanism. Besides, It doesn't use
isochronous packet with CIP header. It uses internal crystal unit as its
initial sampling clock. When detecting input signals which can be
available for sampling clock (e.g. ADAT input), drivers can configure
units to use the signals as source of sampling clock. When something goes
wrong, e.g. frequency mismatching between the signal and configured value,
units fallback to the other detected signals alternatively. When detecting
no alternatives, internal crystal unit is used as source of sampling
clock. On manual of Fireface 400, this mechanism is described as
'Autosync'.
On the units, packet streaming is controlled by write transactions to
certain registers. Format of the packet, e.g. the number of data channels
in a data block, is also configured by the same manner. For this purpose,
.begin_session and .finish_session is added.
The remarkable point of this protocol is to allow drivers to configure
arbitrary sampling transmission frequency; e.g. 12.345 Hz. As long as I
know, there's no actual DAC/ADC chips which support this kind of
capability. I think a pair of packet streaming layer and data block
processing layer is isolated from sampling data processing layer in a
point of governed clock. In short, between these parts, resampling layer
exists. Actually, for Fireface 400, write transactions to
0x'0000'8010'051c has an effect to change sampling clock frequency with
base frequencies (32.0/44.1/48.0 kHz) and its multipliers (x2/x4),
regardless of sampling transmission frequency.
For this reason, the abstraction layer doesn't handle parameters for
sampling clock. Instead, each implementation of .begin_session is
expected to configure sampling transmission frequency.
For packet streaming layer, it's enough to get current selection of
source signals for the sampling clock and its frequency. In the
abstraction layer, when internal crystal is selected, drivers can sets
arbitrary sampling frequency, else they should follow configured
frequency. For this purpose, .get_clock is added.
Drivers are allows to bank up data fetching from a pair of packet
streaming/data block processing layer and sampling data processing layer.
This feature seems to suppress noises at starting/stopping packet
streaming. For this purpose, .switch_fetching_mode is added.
As I described in the above, units have remarkable mechanism to manage
sampling clock and process sampling data. For debugging purpose,
.dump_sync_status and .dump_clock_config are added. I don't have a need
to common interface to represent the status and configuration,
developers can add actual implementation of the abstraction layer as they
like.
Unlike PCM frames, MIDI messages are transferred by asynchronous
communication over IEEE 1394 bus, thus target addresses are important for
this feature. The .midi_high_addr_reg, .midi_rx_port_0_reg and
.midi_rx_port_1_reg are for this purpose. I'll describe them in following
commit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
RME Fireface series has several models and their specifications are
different. Currently, we find no way to retrieve the specifications
from actual devices and need to implement them in this driver.
This commit adds a structure to describe model specific data. This
structure has an identical name for each unit, and maximum number of
data channels in each mode. I'll describe about the mode in following
commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just after appearing on IEEE 1394 bus, this unit generates several bus
resets. This is due to loading firmware from on-board flash memory and
initialize hardware. It's better to postpone sound card registration.
This commit schedules workqueue to process actual probe processing
2 seconds after the last bus-reset. The card instance is kept at unit
probe callback and released at card free callback. Therefore, when the
actual probe processing fails, the memory block is wasted. This is due to
simplify driver implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for RME Fireface series. This commit just
creates/removes card instance according to IEEE 1394 bus event. More
functions will be added in following commits.
Three types of firmware have released by RME GmbH; for Fireface 400, for
Fireface 800 and for UCX/802/UFX. It's reasonable that these models use
different protocol for communication. Currently, I've investigated
Fireface 400 and nothing others.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current ALSA SoC Sound Card basically consists of CPU/Codec/Platform
components. If system uses Kernel modules, we can disable these drivers
by using rmmod command. In such case, we can't disable
CPU/Codec/Platform driver without disabling Sound Card driver.
But on the other hand, we can disable these drivers by using unbind
command. In such case, we can disable these drivers randomly.
In this case, we can create dirty Sound Card which is missing necessary
components.
(1) If user disabled Sound Card first, but did nothing to other drivers,
user can't use Sound because Sound Card is no longer exists.
(2) If user disabled CPU/Codec/Platform driver randomly, but did nothing
to Sound Card, user still be able to use Sound Card, because dirty Sound
Card still exists. In this case, Sound system will be crashed if user
started sound playback/capture. But we can't block such random unbind
now.
To avoid Sound Card crash in (2) case, we need to unregister Sound Card
whenever CPU/Codec/Platform component were unregistered.
This patch solves this issue.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a separate function for deriving (sysclk, lrclk, bclk)
when the clock is auto or pll.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.
But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.
Before this patch:
$ modinfo sound/soc/codecs/snd-soc-rt5677.ko | grep alias
alias: i2c:RT5677CE:00
alias: i2c:rt5676
alias: i2c:rt5677
After this patch:
$ modinfo sound/soc/codecs/snd-soc-rt5677.ko | grep alias
alias: of:N*T*Crealtek,rt5677C*
alias: of:N*T*Crealtek,rt5677
alias: i2c:RT5677CE:00
alias: i2c:rt5676
alias: i2c:rt5677
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.
But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.
Before this patch:
$ modinfo sound/soc/codecs/snd-soc-wm8978.ko | grep alias
alias: i2c:wm8978
After this patch:
$ modinfo sound/soc/codecs/snd-soc-wm8978.ko | grep alias
alias: i2c:wm8978
alias: of:N*T*Cwlf,wm8978C*
alias: of:N*T*Cwlf,wm8978
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.
But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.
Before this patch:
$ modinfo sound/soc/codecs/snd-soc-uda1380.ko | grep alias
alias: i2c:uda1380
After this patch:
$ modinfo sound/soc/codecs/snd-soc-uda1380.ko | grep alias
alias: of:N*T*Cnxp,uda1380C*
alias: of:N*T*Cnxp,uda1380
alias: i2c:uda1380
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.
But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.
Before this patch:
$ modinfo sound/soc/codecs/snd-soc-sta529.ko | grep alias
alias: i2c:sta529
After this patch:
$ modinfo sound/soc/codecs/snd-soc-sta529.ko | grep alias
alias: of:N*T*Cst,sta529C*
alias: of:N*T*Cst,sta529
alias: i2c:sta529
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.
But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.
Before this patch:
$ modinfo sound/soc/codecs/snd-soc-ssm4567.ko | grep alias
alias: acpi*:INT343B:*
alias: i2c:ssm4567
After this patch:
$ modinfo sound/soc/codecs/snd-soc-ssm4567.ko | grep alias
alias: acpi*:INT343B:*
alias: of:N*T*Cadi,ssm4567C*
alias: of:N*T*Cadi,ssm4567
alias: i2c:ssm4567
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver doesn't have a struct of_device_id table but supported devices
are registered via Device Trees. This is working on the assumption that a
I2C device registered via OF will always match a legacy I2C device ID and
that the MODALIAS reported will always be of the form i2c:<device>.
But this could change in the future so the correct approach is to have an
OF device ID table if the devices are registered via OF.
Before this patch:
$ modinfo sound/soc/codecs/snd-soc-rt5645.ko | grep alias
alias: acpi*:10EC3270:*
alias: acpi*:10EC5640:*
alias: acpi*:10EC5650:*
alias: acpi*:10EC5648:*
alias: acpi*:10EC5645:*
alias: i2c:rt5650
alias: i2c:rt5645
After this patch:
$ modinfo sound/soc/codecs/snd-soc-rt5645.ko | grep alias
alias: of:N*T*Crealtek,rt5650C*
alias: of:N*T*Crealtek,rt5650
alias: of:N*T*Crealtek,rt5645C*
alias: of:N*T*Crealtek,rt5645
alias: acpi*:10EC3270:*
alias: acpi*:10EC5640:*
alias: acpi*:10EC5650:*
alias: acpi*:10EC5648:*
alias: acpi*:10EC5645:*
alias: i2c:rt5650
alias: i2c:rt5645
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The driver has an OF device ID table but the struct i2c_driver
.of_match_table field is not set.
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The I2C core always reports a MODALIAS of the form i2c:<foo> even if the
device was registered via OF, this means that exporting the OF device ID
table device aliases in the module is not needed. But in order to change
how the core reports modaliases to user-space, it's better to export it.
While there, move the MODULE_DEVICE_TABLE(i2c, max9867_i2c_id) just next
to the I2C device table declaration, for consistency with other drivers.
Before this patch:
$ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias
alias: i2c:max9867
After this patch:
$ modinfo sound/soc/codecs/snd-soc-max9867.ko | grep alias
alias: i2c:max9867
alias: of:N*T*Cmaxim,max9867C*
alias: of:N*T*Cmaxim,max9867
Signed-off-by: Javier Martinez Canillas <javier@osg.samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Newer ADSP2V2 codecs include a memory protection unit that can
be set to trap illegal accesses. When enabling an ADSPV2 core we
must configure the memory region traps so that the firmware can
access its own memory.
Signed-off-by: Mayuresh Kulkarni <mkulkarni@opensource.wolfsonmicro.com>
Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds support for ADSP2V2 cores. Primary differences are that
they use a 32-bit register map compared to the 16-bit register
map of ADSP2V1, and there are some changes to clocking control.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The new controls will give user the ability to route the left PDM channel
data to the right headset/handsfree DAC.
HS mono to stereo switch: PDM channel 1 (or mono) data to both HS DAC.
HF mono to stereo switch: PDM channel 3 data to both HF DAC.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Apply the same methods to obtain the current stream position as ASoC
Intel SKL driver uses. It reads the position from DPIB for a playback
stream while it still reads from the position buffer for a capture
stream. For a capture stream, some ugly workaround is needed to
settle down the inconsistent position.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The macros _snd_hdac_chip_read() and *_write() expand to different
types (b,w,l) per their argument. They were thought to be used only
internally for other snd_hdac_chip_*() macros, but in some situations
we need to call these directly, and they are way too ugly.
Instead of saving a few lines, we just write these macros explicitly
with the types, so that they can be used in a saner way.
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With the previous unsigned long value clang generates warnings like
this:
sound/pci/hda/patch_ca0132.c:860:37: error: implicit conversion from
'unsigned long' to 'u32' (aka 'unsigned int') changes value from
18446744073709551615 to 4294967295 [-Werror,-Wconstant-conversion]
spec->curr_chip_addx = (res < 0) ? ~0UL : chip_addx;
~ ^~~~
Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "r1" struct has memory holes. We clear it with memset on one path
where it is used but not the other. Let's just memset it at the start
of the function so it's always safe.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We just checked "id.card < 0" on the lines before so we know it's not
true here. We can delete that check.
Also checkpatch.pl complains about some extra curly braces so we may as
well fix that while we're at it.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If we can't fill the "patch" struct because "count" is too small (it can
be as low as 4 bytes) or because copy_from_user() failed, then just
return instead of using unintialized data.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently snd-usb-audio driver received a new option, quirk_alias, to
allow user to apply the existing quirk for a different device. This
works for many quirks as is, but some still need more tune-ups:
namely, some quirks check the USB vendor/device IDs in various places,
thus it doesn't work as long as the ID is different from the expected
one.
With this patch, the driver stores the aliased USB ID, so that these
rest quirks per device ID are applied. The transition to use the
cached USB ID was already done in the past, so what we needed now is
only to overwrite chip->usb_id.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On this Dell AIO machine, the lineout jack does not work.
We found the pin 0x1a is assigned to lineout on this machine, and in
the past, we applied ALC298_FIXUP_DELL1_MIC_NO_PRESENCE to fix the
heaset-set mic problem for this machine, this fixup will redefine
the pin 0x1a to headphone-mic, as a result the lineout doesn't
work anymore.
After consulting with Dell, they told us this machine doesn't support
microphone via headset jack, so we add a new fixup which only defines
the pin 0x18 as the headset-mic.
[rearranged the fixup insertion position by tiwai in order to make the
merge with other branches easier -- tiwai]
Fixes: 59ec4b57bcae ("ALSA: hda - Fix headset mic detection problem for two dell machines")
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
adg is calling of_clk_add_provider() when probe time,
thus, remove should call of_clk_del_provider(), it doesn't now.
This patch fix this issue.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current adg is calling of_clk_add_povider() multiple times,
but it is not correct usage. This patch fixup its parameter
and call it once.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A relatively large pile of fixes for mainline, the first since the merge
window. The biggest block of changes here by volume is the sun8i-codec
set, the driver was newly added in the merge window but it was realized
that renaming some of the user visible controls was required so these
are being pushed for v4.11 to avoid the original code appearing in a
release. Otherwise it's all fairly standard bugfix stuff.
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Merge tag 'asoc-fix-v4.11-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.11
A relatively large pile of fixes for mainline, the first since the merge
window. The biggest block of changes here by volume is the sun8i-codec
set, the driver was newly added in the merge window but it was realized
that renaming some of the user visible controls was required so these
are being pushed for v4.11 to avoid the original code appearing in a
release. Otherwise it's all fairly standard bugfix stuff.
trivial fix to spelling mistake in dev_err error message
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
