As a result of localization of async midi port, ALSA driver for TASCAM
FireWire series can call helper function directly instead of callback
registration.
This commit removes the redundant design.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Linux kernel 4.4, firewire-lib got a feature called as 'async midi port'
for transmission of MIDI message via IEEE 1394 asynchronous communication,
however actual consumer of this feature is ALSA driver for TASCAM FireWire
series only. When adding this feature, I assumed that ALSA driver for
Digi00x might also be a consumer, actually it's not.
This commit moves the feature from firewire-lib to firewire-tascam module.
Two minor kernel APIs are removed.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In kernel APIs of ALSA control interface, drivers can create a control
element set by a call of snd_ctl_new1() with a template. This template
is known to have const qualifier in general cases.
This commit adds the qualifier to template array, for safer program and
runtime. Application of this change moves the symbol from .data section
to .rodata section.
Cc: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An abstraction of asynchronous transaction for transmission of MIDI
messages was introduced in Linux v4.4. Each driver can utilize this
abstraction to transfer MIDI messages via fixed-length payload of
transaction to a certain unit address. Filling payload of the transaction
is done by callback. In this callback, each driver can return negative
error code, however current implementation assigns the return value to
unsigned variable.
This commit changes type of the variable to fix the bug.
Reported-by: Julia Lawall <Julia.Lawall@lip6.fr>
Cc: <stable@vger.kernel.org> # 4.4+
Fixes: 585d7cba5e ("ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd_use_lock_sync() (thus its implementation
snd_use_lock_sync_helper()) has the 5 seconds timeout to break out of
the sync loop. It was introduced from the beginning, just to be
"safer", in terms of avoiding the stupid bugs.
However, as Ben Hutchings suggested, this timeout rather introduces a
potential leak or use-after-free that was apparently fixed by the
commit 2d7d54002e ("ALSA: seq: Fix race during FIFO resize"):
for example, snd_seq_fifo_event_in() -> snd_seq_event_dup() ->
copy_from_user() could block for a long time, and snd_use_lock_sync()
goes timeout and still leaves the cell at releasing the pool.
For fixing such a problem, we remove the break by the timeout while
still keeping the warning.
Suggested-by: Ben Hutchings <ben.hutchings@codethink.co.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Geminilake vendor nid is different from other Skylake variants, but rest
of the initialization code is same.
So a variable is added in hdmi_spec to store the platform specific vendor
nid and move the initialization code to a helper function to be used by
both platform specific init.
Fixes: 126cfa2f5e ("ALSA: hda: Add Geminilake HDMI codec ID")
Signed-off-by: Ander Conselvan De Oliveira <ander.conselvan.de.oliveira@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Jaikrishna Nemallapudi <jaikrishnax.nemallapudi@intel.com>
Cc: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Cc: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Module at the end of DSP pipeline that needs to be connected to a module
in another pipeline are represented as a PGA(leaf node) and in PGA event
handler these modules are bound/unbounded. Modules other than PGA leaf
can be connected directly or via switch to a module in another pipeline.
Example: reference path.
To support the deferred DSP module bind, following changes are done:
o When the path is enabled, the destination module that needs to be
bound may not be initialized. If the module is not initialized, add
these modules in a deferred bind list.
o When the destination module is initialized, check for these modules
in deferred bind list. If found, bind them.
o When the destination module is deleted, Unbind the modules.
o When the source module is deleted, remove the entry from the deferred
bind list.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no need for defining FSL_ESAI_RATES locally as the standard
SNDRV_PCM_RATE_8000_192000 definition can be used instead.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Declare snd_kcontrol_new strcutures as const as they are only passed as
an argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const too.
Done using Coccinelle:
@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};
@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)
@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p
@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Declare snd_kcontrol_new strcutures as const as they are only passed as
an argument to the function snd_ctl_new1. This argument is of type const,
so snd_kcontrol_new structures having this property can be made const too.
Done using Coccinelle:
@r disable optional_qualifier@
identifier x;
position p;
@@
static struct snd_kcontrol_new x@p={...};
@ok@
identifier r.x;
position p;
@@
snd_ctl_new1(&x@p,...)
@bad@
position p != {r.p,ok.p};
identifier r.x;
@@
x@p
@depends on !bad disable optional_qualifier@
identifier r.x;
@@
+const
struct snd_kcontrol_new x;
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
During packet streaming, maximum length of payload for isochronous packet
is invariable, therefore no need to recalculate. Current ALSA IEC 61883-1/6
engine calls a function to calculate it 8,000 or more times per second
for incoming packet processing.
This commit adds a member to have maximum length of payload into 'struct
amdtp_stream', to reduces the function calls. At first callback from
isochronous context, the length is calculated and stored for later
processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Per feedback from Mark Brown, this patch updates the hi6210-i2s
driver to use devm_snd_soc_register_component which simplifies
the logic a bit.
Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Mark Brown suggested a style change to use break in the final
default of a switch statement, so this patch addresses that.
Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch addresses feedback from Mark Brown, adding a few
extra error returns in cases that shouldn't happen
Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
In the case of error in tas2552_codec_probe() we should better
propagate the real error code instead of always returning '-EIO'.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add missing enum texts store in soc_enum.
Signed-off-by: Mousumi Jana <mousumix.jana@intel.com>
Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada@intel.com>
Signed-off-by: Kranthikumar, GudishaX <gudishax.kranthikumar@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Now, we can use .set_jack callback function on codec level. So we
don't need export rt5665_set_jack_detect.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MOTU units transfer/receive messages in each data block of their
isochronous packet payload. A part of content in the message is cleard for
MIDI message transmission, while the rest is unknown yet. Additional
features are required to assist users and developers to reveal the
details.
This commit adds tracepoints for the purpose. The tracepoints are designed
for MOTU's protocol version 2 and 3 (Protocol version 1 is not upstreamed
yet). In the tracepoints, events are probed to gather first two 24 bit
data chunks of each data block. The chunks are formatted into elements
of 64 bit array with padding in MSB.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unique protocol is used for MOTU FireWire series. In this protocol,
data block format is not compliant to AM824 in IEC 61883-1/6. Each of
the data block consists of 24 bit data chunks, except for a first
quadlet. The quadlet is used for source packet header (SPH) described
in IEC 61883-1.
The sequence of SPH seems to represent presentation timestamp
corresponding to included data. Developers have experienced that invalid
sequence brings disorder of units in the series.
Unfortunately, current implementation of ALSA IEC 61883-1/6 engine and
firewire-motu driver brings periodical noises to the units at sampling
transmission frequency based on 44.1 kHz. The engine generates the SPH with
even interval and this mechanism seems not to be suitable to the units.
Further work is required for this issue and infrastructure is preferable
to assist the work.
This commit adds tracepoints for the purpose. In the tracepoints, events
are probed to gather the SPHs from each data blocks.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unique protocol is used for RME Fireface series. In this protocol,
payload format for isochronous packet is not compliant to CIP in
IEC 61883-1/6. The packet includes data blocks just with data channels,
without headers and any metadata.
In previous commits, ALSA IEC 61883-1/6 engine supports this protocol.
However, tracepoints are not supported yet, unlike implementation for
IEC 61883-1/6 protocol. This commit adds support of tracepoints for
the protocol.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce the "lrclk-strength" property to allow LRCLK pad drive strength
to be changed via device tree.
When running a stress playback loop test on a mx6dl wandboard channel
swap can be noticed on about 10% of the times.
While debugging this issue I noticed that when probing the SGTL5000
LRCLK pin with the scope the swap did not happen. After removing
the probe the swap started to happen again.
After changing the LRCLK pad drive strength to the maximum value the
issue is gone.
Same fix works on a mx6dl Colibri board as well.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Max Krummenacher <max.krummenacher@toradex.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There are many codecs with the capability of jack detection. Usually,
we create a jack on machine driver but there is no common function for
machine driver to deliver the jack pointer to codec driver.
snd_soc_codec_set_jack can be used for delivering the jack pointer to
codec driver and enable the jack detection function. To make it work,
codec driver need to define a callback function to receive the jack
pointer and do all necessary procedure for enabling jack detection.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add driver for hi6210 i2s controller found on hi6220 boards.
Signed-off-by: Andy Green <andy.green@linaro.org>
[jstultz: Forward ported to mainline, fairly major rework
based on suggestions from Mark Brown]
Signed-off-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Code can be simplified by using the standard tolower() funtion.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The comment for the FSLSSI_I2S_RATES definition states that the
driver currently only supports I2S slave mode, which is no longer
correct.
As FSLSSI_I2S_RATES is the same as the standard SNDRV_PCM_RATE_CONTINUOUS,
just remove its definition and its comments to make the code simpler.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Another preliminary patch for the dual-codec support: since the
support of vmaster over multiple codecs is difficult, simply disable
it by a new flag to hda_codec struct. A new user hint is added as
well for consistency.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a preliminary patch for a smooth multi-codec support, and it
introduces a new flag, force_pin_prefix, to struct hda_codec.
This flag is used to force to add the pin location prefix to each
input pin. For example, when there is only one microphone pin,
usually the auto-parser assigns the string "Mic". With this flag on,
it'll be like "Front Mic". Also, the creation of "Master" or "PCM"
playback volume for a single pin is suppressed, too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=195305
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On some Intel platforms, the audio clock may not be set correctly
with initial setting. This will cause the audio playback/capture
rates wrong.
This patch checks the audio clock setting and will set it to a
proper value if it is set incorrectly.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=188411
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch refines the definition of AZX_MLCTL_SPA and AZX_MLCTL_CPA
and add more definitions of ML registers
Signed-off-by: Libin Yang <libin.yang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With RTlinux a race condition has been found that leads to NULL ptr crash:
- On CPU 0: uni_player_irq_handler is called to treat XRUN
"(player->state == UNIPERIF_STATE_STOPPED)" is FALSE so status is checked,
dev_err(player->dev, "FIFO underflow error detected") is printed
and then snd_pcm_stream_lock should be called to lock stream for stopping.
- On CPU 1: application stop and close the stream.
Issue is that the stop and shutdown functions are executed while
"FIFO underflow error detected" is printed.
So when CPU 0 calls snd_pcm_stream_lock, player->substream is already null.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When using an external boost supply the PDN_DONE bit is not set, update
the handling in this case to use to use an appropriate fixed delay.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Shorten the time it takes to power down the amp by disabling the volume
ramp whilst doing the final shutdown. The driver has already muted the
amplifier at this stage so doing the volume ramp serves no purpose.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When a matching PLL freq is found, searching continues even this is
not necessary. The problem was introduced with the following refactoring
commit 84fdc00d51 ("ASoC: codec: wm9860: Refactor PLL out freq search)
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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: 5992e30034 ("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 6c29230e2a ("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: 6c29230e2a ("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 c5fcee0373 ("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 9dc5d31cdc ("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: 59ec4b57bc ("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>
We don't need to manually set the card name; with an entry in the
names[] array, this happens automatically.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When DSP module is unbound, the module state needs to be in INIT_DONE
state instead of UNINT. Also the state needs to be set to UNINIT after
module is deleted from DSP pipeline.
So, set the module state to INIT_DONE after unbind and then UNINIT after
module is deleted.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As per hardware recommendation, for every capture stream completion
following operations need to be done in order to reflect the actual
data that is received in position buffer.
1. Wait for 20us before reading the DMA position in buffer once the
interrupt is generated for stream completion.
2. Read any of the register to flush the DMA position value. This is
dummy read operation.
Signed-off-by: Dharageswari R <dharageswari.r@intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skylake driver topology header/driver structure is referenced and used
in SST library which creates circular dependency. Hence the
rearrangement.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We should not hard code the ACPI path to get acpi_handle. Instead use
ACPI_HANDLE macro to do the job.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initially vmixer and mixer widget handlers were bit different, but over
time they became same so remove the duplicate code.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
A module may have multiple instances in DSP, so unload only when usage
count is zero.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add constraint to FE to restrict sample format to 16-bit for bxt_rt298
machine
Signed-off-by: G Kranthi <gudishax.kranthikumar@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When creating the codec dai, use sig_bits to update the max bps based
on the codec capability. So both the link DMA and codec format will be
calculated based on DAI sig_bits.
So update the sig_bits with converter capability and use the sig_bits
for HDA format calculation.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For calculating the HDA DMA format, use the max_bps supported by the
DAI caps instead of fixing it to 32/24. For host DMA the Max bps support
is 32, but in case of link DMA, this depends on the codec capability.
So use the sig_bits to define the bps supported by dai.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Resource managed devm_clk_get only works with platform's device dev.
Reported-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
WM8962 needs its MCLK when powerup in wm8962_resume(). Thus it's better
to control the MCLK in codec driver. Thus remove the clock enable in
machine driver accordingly.
While at it, get rid of imx_wm8962_remove function since it is now
empty.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
LLCH is a 16 bit register. Use readw instead of readl API.
Signed-off-by: B, Jayachandran <jayachandran.b@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current rcar driver is trying to remove kctrl when remove time.
But, 1) rcar driver can't/shouldn't remove before removing sound
card driver, 2) sound card driver will call snd_ctl_dev_free()
and removes all kctrls by snd_ctl_remove().
Thus, rsnd_kctrl_remove() is not necessary. Current implementation
will get Oops when removing rcar driver after sound card.
This patch fix this issue.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add wm8960 machine driver and config option for MT2701.
Signed-off-by: Ryder Lee <ryder.lee@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MOTU 828mk3 (FireWire/Hybrid) is one of third generation in MOTU FireWire
series, produced in 2008/2014. This model consists of three chips for
functionality on IEEE 1394 bus:
* TI TSB41AB2 (Physical layer for IEEE 1394 bus)
* Xilinx Spartan-3E FPGA Family (Link layer for IEEE 1394 bus, packet
processing and data block processing layer)
* TI TMS320C6722 (Digital signal processing)
This commit adds a support for this model, with its unique protocol as
version 3. This protocol has some additional features to protocol
version 2.
* Support several optical interfaces.
* Support a data chunk for return of reverb effect.
* Have a quirk of tx packets.
* Support heartbeat asynchronous transaction.
In this protocol, series of transferred packets has some quirks. Below
fields in CIP headers of the packets are out of IEC 61883-1:
- SID (source node id): always 0x0d
- DBS (data block size): always 0x04
- DBC (data block counter): always 0x00
- EOH (End of header): always 0x00
Below is an actual sample of transferred packets.
quads CIP1 CIP2
520 0x0D040400 0x22FFFFFF
8 0x0D040400 0x22FFFFFF
520 0x0D040400 0x22FFFFFF
520 0x0D040400 0x22FFFFFF
8 0x0D040400 0x22FFFFFF
Status of clock is configured by write transactions to 0x'ffff'f000'0b14,
as well as version 2, while meanings of fields are different from the
former protocols. Modes of optical interfaces are configured by write
transactions to 0x'ffff'f000'0c94.
Drivers can register its address to receive heatbeat transactions from the
unit. 0x'ffff'f000'0b0c is for the higher part and 0x'ffff'f000'0b10 is
for the lower part. Nevertheless, this feature is not useless for this
driver and this commit omits it.
Each data block consists of two parts in a point of the number of included
data chunks. In both of 'fixed' and 'differed' parts, the number of
included data blocks are a multiple of 4, thus depending on models there's
some empty data chunks. For example, 828mk3 includes one pair of empty
data chunks in its fixed part. When optical interface is configured to
S/PDIF, 828mk3 includes one pair of empty data chunks in its differed part.
To reduce consumption of CPU cycles with additional conditions/loops, this
commit just exposes these empty chunks to user space as PCM channels.
Additionally, 828mk3 has a non-negligible overhead to change its sampling
transfer frequency. When softwares send asynchronous transaction to
perform it, LED on the unit starts to blink. In a worst case, it continues
blink during several seconds; e.g. 10 seconds. When stopping blinking,
the unit seems to be prepared for the requested sampling transfer
frequency. To wait for the preparation, this commit forces the driver
to call task scheduler and applications sleeps for 4 seconds.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-1, when two quadlets CIP header is used, the most significant
bit in second CIP header stands. However, packets from units with MOTU
protocol version 3 have a quirk without this flag. Current packet streaming
layer handles this as protocol error.
This commit adds a new enumeration constant for this quirk, to handle MOTU
protocol version 3.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MOTU 828mk2 is one of second generation in MOTU FireWire series, produced in
2003. This model consists of four chips:
* TI TSB41AB2 (Physical layer for IEEE 1394 bus)
* PDI 1394L40BE (Link layer for IEEE 1394 bus and packet processing layer)
* ALTERA ACEX 1K EP1K30 Series FPGA (Data block processing layer)
* TI TMS320VC5402 (Digital signal processing)
This commit adds a support for this model, with its unique protocol as
version 2. The features of this protocol are:
* Support data chunks for status and control messages for both
directions.
* Support a pair of MIDI input/output.
* Support a data chunk for mic/instrument independent of analog line in.
* Support a data chunk for playback return.
* Support independent data chunks for S/PDIF of both optical/coaxial
interfaces.
* Support independent data chunks for each of main out and phone out.
Status of clock is configured by write transactions to 0x'ffff'f000'0b14.
Modes of optical interfaces are configured by write transactions to
0x'ffff'f000'0c04.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MOTU FireWire series can transfer messages to registered address. These
messages are transferred for the status of internal clock synchronization
just after starting streams.
When the synchronization is stable, it's 0x01ffffff. Else, it's 0x05ffffff.
This commit adds a functionality for user space applications to receive
content of the message.
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>
In MOTU FireWire series, MIDI messages are multiplexed to isochronous
packets as well as PCM frames, while the way is different from the one
in IEC 61883-6.
MIDI messages are put into a certain position in message chunks. One data
block can includes one byte of the MIDI messages. When data block includes
a MIDI byte, the block has a flag in a certain position of the message
chunk. These positions are unique depending on protocols.
Once a data block includes a MIDI byte, some following data blocks includes
no MIDI bytes. Next MIDI byte appears on a data block corresponding to
next cycle of physical MIDI bus. This seems to avoid buffer overflow caused
by bandwidth differences between IEEE 1394 bus and physical MIDI bus.
This commit adds MIDI functionality to transfer/receive MIDI messages.
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 according to requests from a userspace application.
Available number of samples in a frame of PCM substream is determined at
open(2) to corresponding PCM character device. Later, packet streaming
starts by ioctl(2) with SNDRV_PCM_IOCTL_PREPARE. In theory, between them,
applications can change state of the unit by any write transaction to
change the number. In this case, this driver may fail packet streaming due
to wrong data format.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to manage packet streaming for MOTU
FireWire series.
The streaming is not controlled by CMP, thus against IEC 61883-1. Write
transaction to certain addresses start/stop packet streaming.
Transactions to 0x'ffff'f000'0b00 results in isochronous channel number for
both directions and starting/stopping transmission of packets. The
isochronous channel number is represented in 6 bit field, thus units can
identify the channels up to 64, as IEEE 1394 bus specification described.
Transactions to 0x'ffff'f000'0b10 results in packet format for both
directions and transmission speed. When each of data block includes fixed
part of data chunks only, corresponding flags stand.
When bus reset occurs, the units continue to transmit packets with
non-contiguous data block counter. This causes discontinuity detection in
packet streaming engine and ALSA PCM applications receives EPIPE from any
I/O operation. In this case, typical applications manage to recover
corresponding PCM substream. This behaviour is kicked much earlier than
callback of bus reset handler by Linux FireWire subsystem, therefore
status of packet streaming is not changed in the handler.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
All models of MOTU FireWire series can be controlled by write transaction
to addresses in a range from 0x'ffff'f0000'0b00 to 0x'ffff'f000'0cff.
The models support asynchronous notification. This notification has 32 bit
field data, and is transferred when status of clock changes. Meaning of
the value is not enough clear yet.
Drivers can register its address to receive the notification. Write
transaction to 0x'ffff'f000'0b04 registers higher 16 bits of the address.
Write transaction to 0x'ffff'f0000'0b08 registers the rest of bits. The
address includes node ID, thus it should be registered every time of bus
reset.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MOTU FireWire series uses blocking transmission for AMDTP packet streaming.
They transmit/receive 8,000 packets per second, to handle the same number
of data blocks as current sampling transmission frequency. Thus,
IEC 61883-1/6 packet streaming engine of ALSA firewire stack is available
for them.
However, the sequence of packet and data blocks includes some quirks.
Below sample is a sequence of CIP headers of packets received by 828mk2,
at 44.1kHz of sampling transmission frequency.
quads CIP1 CIP2
488 0x020F04E8 0x8222FFFF
8 0x020F04F8 0x8222FFFF
488 0x020F0400 0x8222FFFF
488 0x020F0408 0x8222FFFF
8 0x020F04E8 0x8222FFFF
488 0x020F04F0 0x8222FFFF
488 0x020F04F8 0x8222FFFF
The SID (source node ID), DBS (data block size), SPH (source packet header),
FMT (format ID), FDF (format dependent field) and SYT (time stamp) fields
are in IEC 61883-1. Especially, FMT is 0x02, FDF is 0x22 and SYT is 0xffff
to define MOTU specific protocol. In an aspect of dbc field, the value
represents accumulated number of data blocks included the packet. This
is against IEC 61883-1, because according to the specification this value
should be the number of data blocks already transferred.
In ALSA IEC 61883-1/6 engine, this quirk is already supported by
CIP_DBC_IS_END_EVENT flag, because Echo Audio Fireworks has.
Each data block includes SPH as its first quadlet field, to represent its
presentation time stamp. Actual value of SPH is compliant to IEC 61883-1;
lower 25 bits of 32 bits width consists of 13 bits cycle count and 12 bits
cycle offset.
The rest of each data block consists of 24 bit chunks. All of PCM samples,
MIDI messages, status and control messages are transferred by the chunks.
This is similar to '24-bit * 4 Audio Pack' in IEC 61883-6. The position of
each kind of data depends on generations of each model. The number of
whole chunks in a data block is a multiple of 4, to consists of
quadlet-aligned packets.
This commit adds data block processing layer specific for the MOTU
protocol. The remarkable point is the way to generate SPH header. Time
stamps for each data blocks are generated by below calculation:
* Using pre-computed table for the number of ticks per event
* 44,1kHz: (557 + 123/441)
* 48.0kHz: (512 + 0/441)
* 88.2kHz: (278 + 282/441)
* 96.0kHz: (256 + 0/441)
* 176.4kHz: (139 + 141/441)
* 192.0kHz: (128 + 0/441)
* Accumulate the ticks and set the value to SPH for every events.
* This way makes sense only for blocking transmission because this mode
transfers fixed number or none of events.
This calculation assumes that each data block has a PCM frame which is
sampled according to event timing clock. Current packet streaming layer
has the same assumption.
Although this sequence works fine for MOTU FireWire series at sampling
transmission frequency based on 48.0kHz, it is not enough at the frequency
based on 44.1kHz. The units generate choppy noise every few seconds.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit c8bdf49b9935("ALSA: fireworks/firewire-lib: Add a quirk for the
meaning of dbc") adds CIP_DBC_IS_END_EVENT flag just for tx packets.
However, MOTU FireWire series has this quirk for rx packets.
This commit allows both directions with the flag.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-1, CIP headers can have a SPH field. When a packet has 1 in
SPH field of its CIP header, the packet has a source packet headers. A
source packet header consists of 32 bit field (= 1 quadlet) and it
transfers time stamp, which is the same value as the lower 25 bits of the
IEEE 1394 CYCLE_TIMER register and the rest is zero.
This commit just supports source packet header field because IEC 61883-1
includes ambiguity the position of this header and its count. Each
protocol layer is allowed to have actual implementation according its
requirements.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, packet streaming layer passes generated SYT value to data block
processing layer. However, this is not enough in a case that the data block
processing layer generates time stamps by its own ways.
For out-packet stream, the packet streaming layer guarantees 8,000 times
calls of data block processing layers per sec. Therefore, when cycle count
of the first packet is recorded, data block processing layers can calculate
own time stamps with the recorded value.
For the reason, this commit allows packet streaming layer to record the
first cycle count. Each data block processing layer can read the count by
accessing a member of structure for packet streaming layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In an aspect of used protocols to communicate, models of MOTU FireWire
units are categorized to three generations.
This commit adds an abstraction layer of the protocols for features
related to packet streaming functionality. This layer includes 5
operations.
When configuring packet streaming functionality with sampling rate and
sampling transmission frequency, .get_clock_rate and .set_clock_rate are
called with proper arguments. MOTU FireWire series supports up to 192.0kHz.
When checking current source of sampling clock (not clock for packetization
layer), .get_clock_source is used. Enumeration is added to represent the
sources supported by this series. This operation can be used to expose
available sampling rate to user space applications when the unit is
configured to use any input signal as source of clock instead of crystal
clock.
In the protocols, the path between packet processing layer and digital
signal processing layer can be controlled. This looks a functionality to
'mute' the unit. For this feature, .switch_fetching_mode is added. This
can be used to suppress noises every time packet streaming starts/stops.
In a point of the size of data blocks at a certain sampling transmission
frequency, the most units accept several modes. This is due to usage of
optical interfaces. The size differs depending on which modes are
configured to the interfaces; None, S/PDIF and ADAT. Additionally, format
of packet is different depending on protocols. To cache current size of
data blocks and its format, .cache_packet_formats is added. This is used
by PCM functionality, packet streaming functionality and data block
processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
MOTU FireWire series doesn't tell drivers their capabilities, thus
the drivers should have and apply model-dependent parameters to detected
models.
This commit adds a structure to represent such parameters. Capabilities
are represented by enumeration except for the number of analog line
in/out. Identification name also be in the structure because the units has
no registers for this purpose.
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 applies this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds an new driver for MOTU FireWire series. In this commit,
this driver just creates/removes card instance according to bus event.
More functionalities will be added in following commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error path in probe attempts to put the device back into reset.
Should we fail to get the reset_gpio (such as a probe defer) we will
leave the error value in there, which the gpiod_set_value_cansleep on
the error path will attempt to deference.
Fix this issue by clearing reset_gpio before we head into the error
path.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The variable 'data' is assigned null and never re-assigned. There
is also a redundant check for data being non-null which is always
false, so remove this and the variable data and dma_addr as they
are not used once the dead code has been removed.
Detected with CoverityScan, CID#1324015 ("'Constant' variable gaurds
dead code")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Intel SST driver spews an info message "FW Versoin xxxx" at each time
the device gets initialized. Since it's triggered at each PM (or even
runtime PM), it appears so ofetn, and rather becomes annoying than
useful.
This patch suppresses the superfluous messages by checking the
currently loaded FW version with the previously loaded one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
WM8960 derives bit clock from sysclock using BCLKDIV[3:0] of R8
clocking register (See WM8960 datasheet, page 71).
There are use cases, like this:
aplay -Dhw:0,0 -r 48000 -c 1 -f S20_3LE -t raw audio48k20b_3LE1c.pcm
where no BCLKDIV applied to sysclock can give us the exact requested
bitclk, so driver fails to configure clocking and aplay fails to run.
Fix this by relaxing bitclk computation, so that when no exact value
can be derived from sysclk pick the closest value greater than
expected bitclk.
Suggested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
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>
Add a separate function for finding (sysclk, lrclk, bclk)
when the clock is auto or mclk. This makes code easier to
read and reduces the indentation level in wm8960_configure_clocking.
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>
For GPIO-backed pins that are not configured as wakeup sources, we may
miss change in their state that happens while system is suspended. Let's
use PM notifier to refresh their state upon resume.
Signed-off-by: Dmitry Torokhov <dtor@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
R-Car Datasheet is indicating "SSICR.CKDV = 000 is invalid when
SSIWSR.WS_MODE = 1 or SSIWSR.CONT = 1".
Current driver will set CONT, thus, we shouldn't use CKDV = 000.
This patch fixup it.
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>
Declare snd_soc_ops structures as const as they are only stored
in the ops field of a snd_soc_dai_link structure. This field is
of type const, so snd_soc_ops structures having this property
can be made const too.
Cross compiled the .o files for blackfin architecture.
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The clock needs to be stored in the simple_dai structure, so it can
be enabled later on. This has been broken during the conversion to use
devm_* functions for the clk lookup.
Fixes: e984fd61e8 (ASoC: simple-card: use devm_get_clk_from_child())
Signed-off-by: Lucas Stach <l.stach@pengutronix.de>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
reader->substream is used in IRQ handler for error case but is never set.
Set value to pcm substream on DAI startup and clean it on dai shutdown.
Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 1a653aa447 ("ASoC: core: replace aux_comp_list to ...")
tried to replace aux_comp_list to component_dev_list,
but it failed because of binding timing. Thus, Sylwester fixuped it by
commit d2e3a1358c ("ASoC: Fix binding and probing of auxiliary...").
One of main purpose of commit 1a653aa447 ("ASoC: core: replace...")
was remove replaceable list (= list_aux) from snd_soc_component by using
new "auxiliary" flags (but it failed).
Because of this background, current code has reborned card_aux_list
(= same as original list_aux), and almost pointless "auxiliary" flags.
Let's remove pointless "auxiliary" flags by this patch
This means, it is same as revert both
commit 1a653aa447 ("ASoC: core: replace aux_comp_list to ...") and
commit d2e3a1358c ("ASoC: Fix binding and probing of auxiliary...").
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When a new event is queued while processing to resize the FIFO in
snd_seq_fifo_clear(), it may lead to a use-after-free, as the old pool
that is being queued gets removed. For avoiding this race, we need to
close the pool to be deleted and sync its usage before actually
deleting it.
The issue was spotted by syzkaller.
Reported-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The latest gcc-7.0.1 snapshot points out that we if nr_ch is zero, we never
initialize some variables:
sound/pci/au88x0/au88x0_core.c: In function 'vortex_adb_allocroute':
sound/pci/au88x0/au88x0_core.c:2304:68: error: 'mix[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized]
sound/pci/au88x0/au88x0_core.c:2305:58: error: 'src[0]' may be used uninitialized in this function [-Werror=maybe-uninitialized]
I assume this can never happen in practice, but adding a check here doesn't
hurt either and avoids the warning. The code has been unchanged since
the start of git history.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A new Dell laptop needs to apply ALC269_FIXUP_DELL1_MIC_NO_PRESENCE to
fix the headset problem, and the pin definiton of this machine is not
in the pin quirk table yet, now adding it to the table.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a platform clock widget to turn off the clock only when
both headset capture and headset playback are not in use. This removes
turning off the clock in hw_free so that the clock is on when
either capture or playback of headset is in progress.
Signed-off-by: Harsha Priya <harshapriya.n@intel.com>
Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_seq_pool_done() is called, it marks the closing flag to
refuse the further cell insertions. But snd_seq_pool_done() itself
doesn't clear the cells but just waits until all cells are cleared by
the caller side. That is, it's racy, and this leads to the endless
stall as syzkaller spotted.
This patch addresses the racy by splitting the setup of pool->closing
flag out of snd_seq_pool_done(), and calling it properly before
snd_seq_pool_done().
BugLink: http://lkml.kernel.org/r/CACT4Y+aqqy8bZA1fFieifNxR2fAfFQQABcBHj801+u5ePV0URw@mail.gmail.com
Reported-and-tested-by: Dmitry Vyukov <dvyukov@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CONFIG_SND_X86 is a menu config to filter only for x86-specific
drivers in its sub-menu, and this doesn't have to be tristate but
rather it should be a bool. Also, like other sub-menu configs, it's
more user-friendly to be default=y; it's merely a menu config and the
actual drivers are configured in the sub-menu, after all.
Fixes: 287599cf2d ("ALSA: add Intel HDMI LPE audio driver for BYT/CHT-T")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Update the driver to use SND_SOC_DAPM_AIF_IN instead of
SND_SOC_DAPM_DAC.
Rename the interface's widgets to be more precise on which slot
the interface is connected.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
An unwanted space is present in an audio widget's name on the dapm
routing. It causes an error on the recognition of this widget (error:
("no dapm match for AIF1 Slot 0 Right").
Remove the space fixes it.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Update the driver to use the new SOC_DAPM_DOUBLE definition
on the digital DAC mixer.
Update the names accordingly as, when they are shared, the
controls are not prefixed with the widget's name anymore.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The "HP" widget is already present and take part to
the analog part (sun8i-codec-analog).
Remove it from the digital part as it is unnecessary.
Signed-off-by: Mylène Josserand <mylene.josserand@free-electrons.com>
Acked-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The extra pairs of parantheses are not needed and causes clang to
generate warnings like this:
sound/pci/hda/patch_ca0132.c:1171:14: error: equality comparison with extraneous parentheses [-Werror,-Wparentheses-equality]
if ((buffer == NULL))
~~~~~~~^~~~~~~
sound/pci/hda/patch_ca0132.c:1171:14: note: remove extraneous parentheses around the comparison to silence this warning
if ((buffer == NULL))
~ ^ ~
sound/pci/hda/patch_ca0132.c:1171:14: note: use '=' to turn this equality comparison into an assignment
if ((buffer == NULL))
Signed-off-by: Matthias Kaehlcke <mka@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
