Use resource managed API then we can remove snd_dmaengine_pcm_unregister()
and snd_soc_unregister_component() calls in .probe error path and .remove.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Let's leave space for the NUL char otherwise the static checkers
complain that we go beyond the end of the array.
Fixes: 53b3ffee78 ('ALSA: firewire-tascam: change device probing processing')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device has no mixer (and identifies itself as such), so just skip
the mixer definition.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For the Zoom R16/24 (tx_length_quirk set), when calculating the maximum
sample frequency, consideration must be made for the fact that four bytes
of the packet contain a length descriptor and consequently must not be
counted as part of the audio data.
This is corroborated by the wMaxPacketSize for this device, which is 108
bytes according for the USB playback endpoint descriptor. The frame size
is 8 bytes (2 channels of 4 bytes each), and the 108 bytes thus work out
as 13 * 8 + 4, i.e. corresponding to 13 frames plus the additional 4 byte
length descriptor.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)
The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).
In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.
For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.
The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.
In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.
Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.
The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.
Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Tested-by: Panu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireOne is based on OXFW971 and ALSA OXFW driver can support it.
These are values of identical registers.
$ ./firewire-request /dev/fw1 read 0xfffff0050000
result: 97100105
$ ./firewire-request /dev/fw1 read 0xfffff0090020
result: 39373100
This commit adds an entry for this model. This model has physical controls
and its MIDI control messages are transferred to second MIDI data stream
multiplexed in one MIDI conformant data channel.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, sequence multiplexing is applied to MIDI conformant data
channel. As a result, eight MIDI data streams are included in the channel.
Although ALSA AM824 data block processing layer implements this
multiplexing, current OXFW driver doesn't utilize it due to wrong
calculation of MIDI ports.
This commit fixes this bug to add proper calculation. Although this commit
allows to use 8 MIDI data streams, the number of available MIDI ports is
limited by the number of ALSA MIDI ports added by the driver.
Fixes: df075feefbd3('ALSA: firewire-lib: complete AM824 data block processing layer')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Current OXFW driver calculates the number of MIDI ports just before adding
ALSA MIDI ports. It's convenient for some devices with quirks to move
these codes before handling quirks.
This commit implements this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When asynchronous MIDI port is closed before callbacked, the callback
function causes NULL pointer dereference to missing MIDI substream.
This commit fixes this bug.
Fixes: e8a40d9bcb23('ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages')
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>
The contents of Config ROM in firewire device structure are already
aligned to CPU-endianness. Thus, no need to convert it again.
This commit removes needless conversions
Fixes: 9edf723fd858('ALSA: firewire-digi00x: add skeleton for Digi 002/003 family')
Fixes: c0949b278515('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Reported-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, metering is supported for BeBoB based models
customized by M-Audio. The data in transaction is aligned to
big-endianness, while in the driver code u16 typed variable is assigned
to the data. This causes sparse warnings.
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
bebob_maudio.c:651:31: warning: cast to restricted __be16
This commit fixes this bug by using __be16 variable for the data.
Fixes: 3149ac489ff8('ALSA: bebob: Add support for M-Audio special Firewire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, snd_efw_command_get_phys_meters() was added to handle
metering data. The given buffer is used to save transaction result and to
convert between endianness. But this causes sparse warnings.
fireworks_command.c:269:25: warning: incorrect type in argument 1 (different base types)
fireworks_command.c:269:25: expected unsigned int [usertype] *p
fireworks_command.c:269:25: got restricted __be32 [usertype] *
This commit fixes this bug.
Fixes: bde8a8f23bbe('ALSA: fireworks: Add transaction and some commands')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, u32 data was assigned to __be32 variable instead of an
int variable. This is not enough solution because it still causes sparse
warnings.
dice.c:80:23: warning: incorrect type in assignment (different base types)
dice.c:80:23: expected restricted __be32 [usertype] value
dice.c:80:23: got unsigned int
dice.c:81:21: warning: restricted __be32 degrades to integer
dice.c:81:46: warning: restricted __be32 degrades to integer
This commit fixes this bug.
Fixes: 7c2d4c0cf5ba('ALSA: dice: Split transaction functionality into a file')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some local variables in some functions are typed as unsigned int, while
__be32 value is assigned to them. This causes sparse warnings.
dice-stream.c:50:17: warning: incorrect type in assignment (different base types)
dice-stream.c:50:17: expected unsigned int [unsigned] channel
dice-stream.c:50:17: got restricted __be32 [usertype] <noident>
dice-stream.c:74:17: warning: incorrect type in assignment (different base types)
dice-stream.c:74:17: expected unsigned int [unsigned] channel
dice-stream.c:74:17: got restricted __be32 [usertype] <noident>
This commit fixes this bug.
Fixes: 288a8d0cb04f('ALSA: dice: Change the way to start stream')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently everyone and their dog has their own favourite spelling
for vga_switcheroo. This makes it hard to grep dmesg for log entries
relating to vga_switcheroo. It also makes it hard to find related
source files in the tree.
vga_switcheroo.c uses pr_fmt "vga_switcheroo". Use that everywhere.
Signed-off-by: Lukas Wunner <lukas@wunner.de>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Link: http://patchwork.freedesktop.org/patch/msgid/9b0175319ce78d831acfcf11e4c6c760f826b0e3.1444663039.git.lukas@wunner.de
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
A former commit moves oxfw-related codes to a sub-directory, while it
forgot to remove an entry from Makefile in parent directory.
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When committed to upstream, these four modules had wrong entries for
Makefile. This forces them to be loadable modules even if they're set
as built-in.
This commit fixes this bug.
Fixes: b5b04336015e('ALSA: fireworks: Add skelton for Fireworks based devices')
Fixes: fd6f4b0dc167('ALSA: bebob: Add skelton for BeBoB based devices')
Fixes: 1a4e39c2e5ca('ALSA: oxfw: Move to its own directory')
Fixes: 14ff6a094815('ALSA: dice: Move file to its own directory')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent commit [7fbe824a0f0e: ALSA: hda - Update mixer name for the
lower codec address] tried to improve the mixer chip name assignment
in the order of codec address. However, this fix was utterly bogus;
it checks the field set in each codec, thus this value is reset at
each codec creation, of course. For really handling this priority,
the assignment has to be remembered in the common place, namely in
hda_bus, instead of hda_codec.
Fixes: 7fbe824a0f ('ALSA: hda - Update mixer name for the lower codec address')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
My static checker complains because tscm->spec->midi_capture_ports is
either 2 or 4 but the tscm->tx_midi_substreams[] array has 4 elements so
this is possibly off by one. I have looked at the code and I think it
should be >= instead of > as well.
Fixes: 107cc0129a ('ALSA: firewire-tascam: add support for incoming MIDI messages by asynchronous transaction')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We recently tried to add some new code to support turning the LED on and
off but the code in snd_tscm_transaction_reregister() is unreachable.
Fixes: e65e2cb99e ('ALSA: firewire-tascam: Turn on/off FireWire LED')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch rechecks the jack detect status after resuming from S3.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The index_cache is per instance run time state but rt298_index_def is not.
Make rt298_index_def const and make a copy of memory for index_cache rather
than directly use the rt298_index_def.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SX_TLV controls are intended for situations where the register behind
the control has some non-zero value indicating the minimum gain
and then gains increasing from there and eventually overflowing through
zero.
Currently every CODEC implementing these controls specifies the minimum
as the non-zero value for the minimum and the maximum as the number of
gain settings available.
This means when the info callback subtracts the minimum value from the
maximum value to calculate the number of gain levels available it is
actually under reporting the available levels. This patch fixes this
issue by adding a new snd_soc_info_volsw_sx callback that does not
subtract the minimum value.
Fixes: 1d99f2436d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV")
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Tested-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Probing from Gen1 is not error. This patch fixup it
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Some of the default value on rt298_index_def are incorrect. Change
them to the correct value.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
PCM timer is not always used. For embedded device, we need an interface
to disable it when it is not needed, to shrink the kernel size and
memory footprint, here add CONFIG_SND_PCM_TIMER for it.
When both CONFIG_SND_PCM_TIMER and CONFIG_SND_TIMER is unselected,
about 25KB saving bonus we can get.
Please be noted that when disabled, those stubs who using pcm timer
(e.g. dmix, dsnoop & co) may work incorrectlly.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jie Yang <yang.jie@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Nocturn needs the MIDI_RAW_BYTES quirk, like other Novation devices.
Tested that the Nocturn shows up in aconnect, and that it can be used
as a control surface (using the xtor synthesizer patch editor).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In most cases, we prefer the onboard codec as the primary device, thus
it's better to set it as the mixer name. Currently, however, the
mixer name is updated per the device instantiation order, and user
gets often HDMI/DP or other seen as a mixer chip name. Also, if a
codec name is renamed by the driver, the old chip name might be left
still as the mixer name.
This patch addresses these issues by remembering the chip address that
was referred as the mixer name. When a codec with the same or lower
address gives its name, renew the mixer name accordingly, as it's
either the update of the codec name or we get likely the more
appropriate chip as the reference.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A few multiple codec drivers do renaming the chip_name string but all
these are open-coded and some of them have even no error check. Let's
make common helpers to do it properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cirrus codecs have also fine power controls on each widget, thus it
gets benefit from the recent widget power-saving feature. As we
haven't seen any obvious regressions with tests on some MacBooks,
let's try to enable it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We cap the upper bound of "idx" but not the negative side. Let's make
it unsigned to fix this.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Gen1 SRU support was created for preparation of Gen2 SRC support,
but no-one is using this feature (sampling rate convert) on Gen1.
BockW had used SRU before, but it was pass through mode.
This means it is same as SSI. And BockW "platform base" code was
removed from upstream code. It is now supported via DT, but it doesn't
use SRU. More detail, r8a7778.dtsi has "rcar_sound,src" entry, but
no-one is using this feature today. SRU probing has no relation to this
removing. This means there is no effect for DT compatibility, no issues
on upstream kernel.
Gen2 SRC was created from Gen1 SRU, these are similar but not same IP.
Keeping Gen1 SRU in current driver is a little bit difficult,
and no-one is using it today. Gen1 sound is still supported via SSI.
Gen1 SRU support will be removed in the next kernel version.
This patch announces it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.
We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.
Detailed explanation and rationale:
The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:
maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
>> (16 - ep->datainterval);
Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.
The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.
In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.
The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.
Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).
This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.
The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.
For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.
Rephrasing the maxsize expression to:
maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
(frame_bits >> 3);
for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.
We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):
Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56
This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .
(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)
Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Compiling the hdac extended core on arm fails with below error:
sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_writel':
>> sound/hda/ext/hdac_ext_bus.c:29:2: error: implicit declaration of
>> function
+'writel' [-Werror=implicit-function-declaration]
writel(value, addr);
^
sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_readl':
>> sound/hda/ext/hdac_ext_bus.c:34:2: error: implicit declaration of
>> function
+'readl' [-Werror=implicit-function-declaration]
return readl(addr);
This is fixed by explicitly including io.h
Fixes: 99463b3a39 - ('ALSA: hda: provide default bus io ops extended hdac')
Reported-by: kbuild test robot <lkp@intel.com>
Suggested-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It is not necessary to set registers volatile. So, return false
for default case of rt298_volatile_register.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, this driver picks up model name with be32_to_cpu() macro
to align characters. This is wrong operation because the result is
different depending on CPU endiannness.
Additionally, vendor released several versions of firmware for this
series. It's not better to assign model-dependent information to
device entry according to the version field.
This commit fixes these bugs. The name of model is picked up correctly
and used to identify model-dependent information.
Cc: Stefan Richter <stefanr@s5r6.in-berlin.de>
Fixes: c0949b2785 ('ALSA: firewire-tascam: add skeleton for TASCAM FireWire series')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series has some LEDs on its surface. These LEDs can be
turned on/off by receiving asynchronous transactions to a certain
address. One of the LEDs is labels as 'FireWire'. It's better to light it
up when this driver starts to work. Besides, the LED for 'FireWire' is
turned off at bus reset.
This commit implements this idea.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, this driver got functionalities to transfer/receive
MIDI messages to/from TASCAM FireWire series.
This commit adds some ALSA MIDI ports to enable userspace applications
to use the functionalities.
I note that this commit doesn't support virtual MIDI ports which console
models support. A physical controls can be assigned to a certain MIDI
ports including physical and virtual. But the way is not clear.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series use asynchronous transaction to receive MIDI
messages. The transaction should be sent to a certain address.
This commit supports the outgoing MIDI messages. The messages in the
transaction includes some quirks:
* One MIDI message is transferred in one quadlet transaction, except for
system exclusives.
* MIDI running status is not allowed, thus transactions always include
status byte.
* The basic data format is the same as transferring MIDI messages
supported in previous commit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series use asynchronous transaction to transfer MIDI
messages. The transaction is sent to a registered address.
This commit supports the incoming MIDI messages. The messages in the
transaction include some quirks:
* Two quadlets are used for one MIDI message and one timestamp.
* Usually, the first byte of the first quadlet includes MIDI port and MSB
4 bit of MIDI status. For system exclusive message, the first byte
includes MIDI port and 0x04, or 0x07 in the end of the message.
* The rest of the first quadlet includes MIDI bytes up to 3.
* Several set of MIDI messages and timestamp can be transferred in one
block transaction, up to 8 sets.
I note that TASCAM FireWire series ignores ID bytes of system exclusive
message. When receiving system exclusive messages with ID bytes on physical
MIDI bus, the series transfers the messages without ID bytes on IEEE 1394
bus, and vice versa.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commits, asynchronous transactions are supported for physical
controls. This commit adds a pair of MIDI ports for them.
This driver already adds diferrent number of ALSA MIDI ports for physical
MIDI ports, and the number of in/out ports are different. As seeing as
'amidi' program in alsa-utils package, a pair of in/out MIDI ports is
expected with the same name. Therefore, this commit adds a pair of new
ports to the first.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, asynchronous transaction for incoming MIDI messages
from physical controls is supported. The physical controls may be
controlled by receiving MIDI messages at a certain address.
This commit supports asynchronous transaction for this purpose.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 00x series has two types of model; rack and console. The console
models have physical controls. The model can transmit control messages.
These control messages are transferred by asynchronous transactions to
registered address.
This commit supports the asynchronous transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds MIDI functionality to capture/playback MIDI messages
from/to physical MIDI ports. These messages are transferred in isochronous
packets.
When no substreams request AMDTP streams to run, this driver starts the
streams at current sampling rate. When other substreams start at different
sampling rate, the streams are stopped temporarily, then start again at
requested sampling rate. This operation can generate missing MIDI bytes,
thus it's preferable to start PCM substreams at favorite sampling rate in
advance.
Digi 002/003 console also has a set of MIDI port for physical controls.
These ports are added in later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In Digi 002/003 protocol, MIDI messages are transferred in the last data
channel of data blocks. Although this data channel has a label of 0x80,
it's not fully MIDI conformant data channel especially because the Counter
field always zero independently of included MIDI bytes. The 4th byte of
the data channel in LSB tells the number of included MIDI bytes. This byte
also includes the number of MIDI port. Therefore, the data format in this
data channel is:
* 1st: 0x80 as label
* 2nd: MIDI bytes
* 3rd: 0 or MIDI bytes
* 4th: the number of MIDI byte and the number of MIDI port
This commit adds support of MIDI messages in data block processing layer.
Like AM824 data format, this data channel has a capability to transfer
more MIDI messages than the capability of phisical MIDI bus. Therefore, a
throttle for data rate is required to prevent devices' internal buffer to
overflow.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Original code for 'DoubleOhThree' encoding was written with '__u8' type,
while the type is usually used to export something to userspace.
This commit replaces the type with 'u8'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch enables interrupt transfer mode for MIDI ports on newer
Boss/Roland devices such as the GT-100/001 which support interrupt
transfer on both IN and OUT MIDI endpoints. Previously this wasn't being
enabled for these devices as the code was specifically looking for the
scenario where the IN endpoint supported interrupt transfer and the OUT
endpoint was bulk transfer. Newer devices support interrupt transfer for
both endpoints.
This has been tested on Boss devices GT-001, BR-80 and JS-8 and Roland
VS-20.
It would benefit from some regresison testing with other devices if
possible.
Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In firewire-lib, isochronous packet streaming is stopped when detecting
wrong value for FMT field of CIP headers. Although this is appropriate
to IEC 61883-1 and 6, some BeBoB based devices with vendors' customization
use invalid value to FMT field of CIP headers in the beginning of
streaming.
$ journalctl
snd-bebob fw1.0: Detect unexpected protocol: 01000000 8000ffff
I got this log with M-Audio FireWire 1814. In this line, the value of FMT
field is 0x00, while it should be 0x10 in usual AMDTP.
Except for the beginning, these devices continue to transfer packets with
valid value for FMT field, except for the beginning. Therefore, in this
case, firewire-lib should continue to process packets. The former
implementation of firewire-lib performs it.
This commit loosens the handling of wrong value, to continue packet
processing in the case.
Fixes: 414ba022a5 ('ALSA: firewire-lib: add support arbitrary value for fmt/fdf fields in CIP header')
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The structures of type snd_bebob_clock_spec, snd_bebob_rate_spec,
snd_bebob_meter_spec, and snd_bebob_spec are never modified after they are
initialized. Make them all const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This ensures that the link is not requesting any clock and the
PLL can turn off. The link is powered when controller is brought
out of reset.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
On runtime pm resume, we need to download the firmware, also on
suspend we need to ensure all the interrupts from controller and
DSP are disabled.
Also since we download the firmware on resume, we don't need to do
so on init, so remove that bit
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Like we have in legacy mode HDA driver, we need to check the
status bit and handle interrupt only when it is not zero or all
bits set. We typically see the status as all 1's when controller
resumes from suspend, So add the check here as well and don't
handle for these cases.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Skylake driver will set the SPA bit to 0 to turn off the DSP core.
Driver will poll the Current Power Active (CPA) bit to match the
Set Power Active (SPA) bit value. When CPA bit matches the value
of SPA bit, the achieved power state has reached.
In case of DSP power down, register that was polled is SPA
instead of CPA. This patch corrects the register to be polled
in case of DSP power down.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently, when asynchronous transactions finish in error state and
retries, work scheduling and work running also continues. This
should be canceled at fatal error because it can cause endless loop.
This commit enables to cancel transferring MIDI messages when transactions
encounter fatal errors. This is achieved by setting error state.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Typically, the target devices have internal buffer to adjust output of
received MIDI messages for MIDI serial bus, while the capacity of the
buffer is limited. IEEE 1394 transactions can transfer more MIDI messages
than MIDI serial bus can. This can cause buffer over flow in device side.
This commit adds throttle to limit MIDI data rate by counting intervals
between two MIDI messages. Usual MIDI messages consists of two or three
bytes. This requires 1.302 to 1.953 mili-seconds interval between these
messages. This commit uses kernel monotonic time service to calculate the
time of next transaction.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when two MIDI trigger callbacks can be called immediately,
transactions for the second MIDI messages can be postpone till next trigger
callback. This is not good for real-time message transmission.
This commit schedules work again at response handling callback if the
MIDI substream still includes untransferred MIDI messages.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, when waiting for a response, callers can start another
transaction by scheduling another work. This is not good for error
processing of transaction, especially the first response is too late.
This commit serialize request/response transactions, by adding one
boolean member to represent idling state.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some models receive MIDI messages via IEEE 1394 asynchronous transactions.
In this case, MIDI messages are transferred in fixed-length payload. It's
nice that firewire-lib module has common helper functions.
This commit implements this idea. Each driver adds
'struct snd_fw_async_midi_port' in its instance structure. In probing,
it should call snd_fw_async_midi_port_init() to initialize the
structure with some parameters such as target address, the length
of payload in a transaction and a pointer for callback function
to fill the payload buffer. At 'struct snd_rawmidi_ops.trigger()'
callback, it should call 'snd_fw_async_midi_port_run()' to start
transactions. Each driver should ensure that the lifetime of MIDI
substream continues till calling 'snd_fw_async_midi_port_finish()'.
The helper functions support retries to transferring MIDI messages when
transmission errors occur. When transactions are successful, the helper
functions call 'snd_rawmidi_transmit_ack()' internally to consume MIDI
bytes in the buffer. Therefore, Each driver is expected to use
'snd_rawmidi_transmit_peek()' to tell the number of bytes to transfer to
return value of 'fill' callback.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_seq_oss_readq_put_event() seems to be missing a memory barrier which
might cause the waker to not notice the waiter and miss sending a
wake_up as in the following figure.
snd_seq_oss_readq_put_event snd_seq_oss_readq_wait
------------------------------------------------------------------------
/* wait_event_interruptible_timeout */
/* __wait_event_interruptible_timeout */
/* ___wait_event */
for (;;) { prepare_to_wait_event(&wq, &__wait,
state);
spin_lock_irqsave(&q->lock, flags);
if (waitqueue_active(&q->midi_sleep))
/* The CPU might reorder the test for
the waitqueue up here, before
prior writes complete */
if ((q->qlen>0 || q->head==q->tail)
...
__ret = schedule_timeout(__ret)
if (q->qlen >= q->maxlen - 1) {
memcpy(&q->q[q->tail], ev, sizeof(*ev));
q->tail = (q->tail + 1) % q->maxlen;
q->qlen++;
------------------------------------------------------------------------
There are two other place in sound/core/seq/oss/ which have similar
code. The attached patch removes the call to waitqueue_active() leaving
just wake_up() behind. This fixes the problem because the call to
spin_lock_irqsave() in wake_up() will be an ACQUIRE operation.
I found this issue when I was looking through the linux source code
for places calling waitqueue_active() before wake_up*(), but without
preceding memory barriers, after sending a patch to fix a similar
issue in drivers/tty/n_tty.c (Details about the original issue can be
found here: https://lkml.org/lkml/2015/9/28/849).
Signed-off-by: Kosuke Tatsukawa <tatsu@ab.jp.nec.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that we have introduced the core fns we should make hda use these
helpers
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current codec helpers are local to hda code and needs to be moved to
core so that other users can use it.
The helpers to read/write the codec and to check the
power state of widgets is copied
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We were getting build warning about "Section mismatch".
dmi_platform_intel_broadwell is being referenced from the probe function
rt5645_i2c_probe(), but dmi_platform_intel_broadwell was marked with
__initdata.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Reviewed-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a driver for the SPDIF transceiver available on RK3066, RK3188 and
RK3288. Heavily based on the rockchip i2s driver.
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
dev_info is too noisy for tplg wiget loading, so move it to
debug level
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Quite a few fixes here but they're all very small and driver specific,
none of them really stand out if you aren't using the relevant hardware
but they're all useful if you do happen to have an affected device.
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Merge tag 'asoc-fix-v4.3-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.3
Quite a few fixes here but they're all very small and driver specific,
none of them really stand out if you aren't using the relevant hardware
but they're all useful if you do happen to have an affected device.
Load and Initialize Non HDA Link Table in Skylake driver
to get platform configuration.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Initialize and creates DSP controls if processing pipe capability
is supported by HW. Updates the dma_id, hw_params to module param
to be used when DSP module has to be configured.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The SKL driver does not code DSP topology in driver. It uses the
newly added ASoC topology core to parse the topology information
(controls, widgets and map) from topology binary.
Each topology element passed private data which contains
information that driver used to identify the module instance
within firmware and send IPCs for that module to DSP firmware
along with parameters.
This patch adds init routine to invoke topology load and callback
for topology creation.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For FE and BE, the PCM parameters come from FE and BE hw_params
values passed. For a FE we convert the FE params to DSP expected
module format and pass to DSP. For a BE we need to find the
gateway settings (i2s/PDM) to be applied. These are queried from
NHLT table and applied.
Further for BE based on direction the settings are applied as
either source or destination parameters.
These helpers here allow the format to be calculated and queried
as per firmware format.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Skylake driver topology model tries to model the firmware
rule for pipeline and module creation.
The creation rule is:
- Create Pipe
- Add modules to Pipe
- Connect the modules (bind)
- Start the pipes
Similarly destroy rule is:
- Stop the pipe
- Disconnect it (unbind)
- Delete the pipe
In driver we use Mixer, as there will always be ONE mixer in a
pipeline to model a pipe. The modules in pipe are modelled as PGA
widgets. The DAPM sequencing rules (mixer and then PGA) are used
to create the sequence DSP expects as depicted above, and then
widget handlers for PMU and PMD events help in that.
This patch adds widget event handlers for PRE/POST PMU and
PRE/POST PMD event for mixer and pga modules. These event
handlers invoke pipeline creation, destroy, module creation,
module bind, unbind and pipeline bind unbind
Event handler sequencing is implement to target the DSP FW
sequence expectations to enable path from source to sink pipe for
Playback/Capture.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
To configure a module, driver needs to send input and output PCM
params for a module in DSP. The FE PCM params come from hw_params
ie from user, for a BE they also come from hw_params but from
BE-link fixups.
So based on PCM params required driver has to find a converter
module (src/updown/format) and then do the conversion and
calculate PCM params in these pipelines In this patch we add the
helper modules which allow driver to do these calculations.
Signed-off-by: Hardik T Shah <hardik.t.shah@intel.com>
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
SKL driver needs to instantiate pipelines and modules in the DSP.
The topology in the DSP is modelled as DAPM graph with a PGA
representing a module instance and mixer representing a pipeline
for a group of modules along with the mixer itself.
Here we start adding building block for handling these. We add
resource checks (memory/compute) for pipelines, find the modules
in a pipeline, init modules in a pipe and lastly bind/unbind
modules in a pipe These will be used by pipe event handlers in
subsequent patches
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Driver now can make use of mclk data, if provided, to set, enable
and disable the clock source. As part of this, the choice to
enable clock squaring is dealt with as part of dai_sysclk() call
rather than as platform data.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support for DT bindings in the codec driver.
As part of this support, the mclk data can now be provided and
used to control the mclk during codec operation.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Pull in the i915/hda changes for N/CTS setting so I can apply the
follow-up documentation work for drm/i915.
Some conflicts because ofc we had to rework i915 while that N/CTS work
was going on. But not more than adjacent changes really.
Signed-off-by: Daniel Vetter <daniel.vetter@intel.com>
Since commit 3d7608e4c1 ("ARM: shmobile: bockw: remove legacy
board file and config"), Renesas R-Car SoCs are only supported in
generic DT-only ARM multi-platform builds. The driver doesn't need to
use platform data anymore, hence remove platform data configuration.
Move <sound/rcar_snd.h> to sound/soc/sh/rcar/, as it's no longer needed
by platform code.
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@kernel.org>
Should only try to enable/disable the provided mclk, during bias
level changes, if it's not NULL. Also return value of
clk_prepare_enable() should be checked and dealt with accordingly.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use of_match_ptr() to handle non-DT kernel scenario where match
table should be NULL.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the boards have their headphone jack directly connected to the
matching pins of the SoCs. Since most of the time we will have the same
routing path, it makes no sense to put that in the DTS, since it will only
be some useless duplication there.
It also fixes the following warning messages that were seen so far, on
boards where we were using the bindings in the documentation example.
sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Left -> direct -> Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: no sink widget found for Headphone Jack
sun4i-codec 1c22c00.codec: ASoC: Failed to add route HP Right -> direct -> Headphone Jack
Reported-by: Priit Laes <plaes@plaes.org>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/codecs/nau8825.c:1096:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
CC: Anatol Pomozov <anatol.pomozov@gmail.com>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hardconding a platform data for dw_dmac let's use it's own
autoconfiguration feature. Thus, remove hardcoded values.
Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make a copy of memory for index_cache rather than directly use the
rt286_index_def to avoid run time error.
Fixes: c418a84a8c ("ASoC: Constify reg_default tables")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
AD1939 is missed from the table, so add it.
AD1936 and AD1937 are controlled by I2C interface, so remove them.
Fixes: e5224f58e3 ("ASoC: ad193x: add support to ad1934")
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
The minimum volume level for the TAS2552 (control register value 0x00)
is -7dB however the driver declares it as -0.07dB.
Running amixer before the patch reports:
dBscale-min=-0.07dB,step=1.00dB,mute=0
Running amixer with the patch applied reports:
dBscale-min=-7.00dB,step=1.00dB,mute=0
Signed-off-by: Andreas Dannenberg <dannenberg@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Change the rockchip i2s object name (and thus module name) from the
rather generic snd-soc-i2s to the more specific snd-soc-rockchip-i2s
Signed-off-by: Sjoerd Simons <sjoerd.simons@collabora.co.uk>
Signed-off-by: Mark Brown <broonie@kernel.org>
The AD1934 codec has no ADC feature. Hence it register mapping is slightly
different from the register mapping of other members of the AD193x family.
Some ASoC controls and widgets are related to the DAC feature so are not
relevant in the case of an AD1934 codec.
Signed-off-by: Cyrille Pitchen <cyrille.pitchen@atmel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, ESAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power
source of modules during system suspend and resume procedure. Thus,
SSI needs to save all the values of registers before the system
suspend and restore them after the system resume.
The register SFCSR is volatile, but some bits in it need to be
recovered after suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SPDIF needs to save
all the values of registers before the system suspend and restore them after
the system resume.
The SRPC register should be volatile, LOCK bit is set by the hardware.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of
modules during system suspend and resume procedure. Thus, SAI needs to save
all the values of registers before the system suspend and restore them after
the system resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add snd_soc_pm_ops in machine driver to make the trigger suspend/resume
be called in suspend/resume.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
dw i2s controller can work in slave mode, codec being master.
dw i2s is made to support master/slave operation, by reading dwc
register.
Signed-off-by: Maruthi Bayyavarapu <maruthi.bayyavarapu@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
if the stream is decoupled and both link and host are used, while
releasing the stream, need to check if link and host stream are
not in use. This patch adds fix to check if the host/link stream
is in used before coupling it back when releasing the stream.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bits in LOSIDV need to be set to map the stream id for specific link.
Fixing this by setting the required bits in the register.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently if bulk_enable() of supplies fails, the code still goes on
to try and put the device into active state, and set expected IO
voltage of the device. This doesn't really make sense so code now
returns on bulk_enable() failure, with an error message. Also,
to help with debug, failure to get supplies also provides an error
message.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When OSS emulation is loaded on ISA SB AWE32 chip, we get now kernel
warnings like:
WARNING: CPU: 0 PID: 2791 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x80()
sysfs: cannot create duplicate filename '/devices/isa/sbawe.0/sound/card0/seq-oss-0-0'
It's because both emux synth and opl3 drivers try to register their
OSS device object with the same static index number 0. This hasn't
been a big problem until the recent rewrite of device management code
(that exposes sysfs at the same time), but it's been an obvious bug.
This patch works around it just by using a different index number of
emux synth object. There can be a more elegant way to fix, but it's
enough for now, as this code won't be touched so often, in anyway.
Reported-and-tested-by: Michael Shell <list1@michaelshell.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds the codec reset setting in the shutdown function to prevent
the weird sound of the headphone happened by rebooting.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The recent widget power saving introduced some unavoidable click
noises on old IDT 92HD73xx chips while it still seems working on the
compatible new chips. In the bugzilla, we tried lots of tests and
workarounds, but they didn't help much. So, let's disable the feature
for these specific chips as the least (but safest) fix.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=104981
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Its a bit odd that debugfs_create_bool() takes 'u32 *' as an argument,
when all it needs is a boolean pointer.
It would be better to update this API to make it accept 'bool *'
instead, as that will make it more consistent and often more convenient.
Over that bool takes just a byte.
That required updates to all user sites as well, in the same commit
updating the API. regmap core was also using
debugfs_{read|write}_file_bool(), directly and variable types were
updated for that to be bool as well.
Signed-off-by: Viresh Kumar <viresh.kumar@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
The MacBookPro 12,1 has the same setup as the 11 for controlling the
status of the optical audio light. Simply apply the existing workaround
to the subsystem ID for the 12,1.
[sorted the fixup entry by tiwai]
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=105401
Signed-off-by: John Flatness <john@zerocrates.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Much like all the other Lenovo laptops, add a quirk to make
sound work with docking.
Reported-and-tested-by: lacknerflo@gmail.com
Signed-off-by: Laura Abbott <labbott@fedoraproject.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codec supports 4 channel recording with TDM on AIF1.
This patch modifies the DAI capability to allow it.
Signed-off-by: Ben Zhang <benzh@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds support for the DA7219 audio codec with built-in advanced
accessory detect features.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Designware I2S uses tx empty and rx available signals as the DMA
handshaking signals. during music playing, if XRUN occurs,
i2s_stop() function will be executed and both tx and rx irq are
masked, when music continues to be played, i2s_start() is executed
but both tx and rx irq are not unmasked which cause I2S stop
sending DMA handshaking signal to DMA controller, and it finally
causes music playing will be stopped once XRUN occurs for the first
time.
[On list discussion suggests this may be partly a race condition on slow
systems -- broonie]
Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The struct snd_soc_tplg_pcm_dai is renamed to snd_soc_tplg_pcm.
This struct will now be used to handle data related to PCMs
(FE DAI & DAI links). It's not for BE, because BE DAI mappings will be
provided by ACPI/FDT data.
Remove the unused struct snd_soc_tplg_pcm_cfg_caps. We are using
snd_soc_tplg_stream and snd_soc_stream_caps instead.
Bump ABI version to 4.
Signed-off-by: Vedang Patel <vedang.patel@intel.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds convenience defines for declaring a gain control that
has an input mux. These blocks are functionally equivalent to
the existing mixer blocks but can only have a single input
active at once.
Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com>
Acked-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit adds hwdep interface so as the other IEEE 1394 sound devices
has.
This interface is designed for mixer/control applications. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds PCM functionality to transmit/receive PCM samples.
When one of PCM substreams are running or external clock source is
selected, current sampling rate is used. Else, the sampling rate is
changed as an userspace application requests.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds streaming functionality for both direction. To utilize
the sequence of the number of data blocks in packets, full duplex with
synchronization is applied.
Besides, TASCAM FireWire series allows drivers to decide which PCM data
channels are enabled. For convenience, this driver always enable whole the
data channels.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series uses non-blocking transmission for AMDTP packet
streaming, while the format of data blocks is unique.
The CIP headers includes specific value in FMT field and no SYT
information.
In transmitted packets, the first data channel represents event counter,
and the last data channel has status and control information. The rest
has 24bit PCM samples with right padding.
In received packets, all of data channels include 16, 24, 32bit PCM
samples. There's no other kind of information.
This commit adds support for this protocol. For convenience, the size of
PCM samples in outgoing packet is limited by 16 and 24bit. The status and
control information will be supported in future commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series has certain registers for firmware information.
This commit adds proc node to show the information.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
TASCAM FireWire series doesn't tell drivers their capabilities, thus
the drivers should have model-dependent parameters and apply it to
detected devices.
This commit adds a structure to represent such parameters.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for TASCAM FireWire series. In this commit,
this driver just creates/removes card instance according to bus event.
More functionalities will be added in following commits.
TASCAM FireWire series consists of:
* PDI 1394P23 for IEEE 1394 PHY layer
* PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
* XILINX XC9536XL
* XILINX Spartan-II XC2S100
* ATMEL AT91M42800A
Ilya Zimnovich had investigated TASCAM FireWire series in 2011, and
discover some features of his FW-1804. You can see a part of his research
in FFADO project.
http://subversion.ffado.org/wiki/Tascam
A part of my work are based on Ilya's investigation, while this series
doesn't support the FW-1804, because of a lack of config ROM
information and its protocol detail, especially for PCM channels.
I observed that FW-1884 and FW-1082 don't work properly with 1394 OHCI
controller based on VT6315. The controller can actually communicate packets
to these models, while these models generate no sounds. It may be due to
the PHY/LINK layer issues. Using 1394 OHCI controller produced by the other
vendors such as Texas Instruments may work. Or adding another node on the
bus.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
from Designware I2S datasheet, tx/rx XRUN irq is cleared by
reading register TOR/ROR, rather than by writing into them.
Signed-off-by: Yitian Bu <yitian.bu@tangramtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
The current code writes a set of registers that are reserved on the
tlc320aic3104. The change skips those registers for that IC.
Signed-off-by: Rick Mann <rmann@latencyzero.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch removes the old PXA DMA API usage and switches over to
generic functions provided by snd-soc-dmaengine-pcm.
More cleanups may be done on top of this, and some function stubs can
now be removed completetly. However, the intention here was to keep
the transition as small as possible.
This was tested on the mioa701 pxa27x board.
Signed-off-by: Daniel Mack <zonque@gmail.com>
[trivial change from mmp-dma to pxa-dma]
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' into asoc-pxa
ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
The SND_PCM_RATE_KNOT covers all the rate settings, even though some that
we don't support, while we also list all the rate we support. Simply remove
it.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The current code, disregarding the clk_set_rate error code, was always
returning -EINVAL. Fix that and return the code in order to have more clue
about what's going on.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
pm_runtime_enable is called in probe to enable runtime PM
for wm8962 codec, but pm_runtime_disable isn't called in remove
callback, nor is called in error path if probe fails after runtime
PM is enabled, this causes unbalanced pm_runtime_enable.
This patch Adds pm_runtime_disable in remove callback and error path,
to balance pm_runtime_enable.
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Digi 002/003 family uses asynchronous transaction for messaging.
The address to transmit this message is stored on a certain register.
This commit allocates a range of address on OHCI 1394 host controller
to handle the messaging. As long as I know, the purpose of this message
seems to notify lost of synchronization. While, the meaning of content
of the message is not clear.
Actual examples of this messaging:
* When clock source is set as internal:
- 0x00007051
- 0x00007052
- 0x00007054
- 0x00007057
- 0x00007058
* When clock source is set as somewhat external:
- 0x00009000
- 0x00009010
- 0x00009020
- 0x00009021
- 0x00009022
The lost often occurs when using internal clock source. In this case,
users hear sounds with quite short gap every several minutes. In fact,
the lost is recovered temporarily.
When using with external clock source, the lost seems not to occur. The
mechanism is not clear yet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds hwdep interface so as the other sound drivers for units
on IEEE 1394 bus have.
This interface is designed for mixer/control applications. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds PCM functionality to transmit/receive PCM samples.
Any PCM substreams are jointed because incoming/outgoing AMDTP streams
are bound. When one of PCM substream is running or external clock source
is selected, current sampling rate is used. Else, the sampling rate is
changed as an userspace application requests.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds proc node to show current clock status for debugging.
As long as testing Digi 002 rack, registers can show local clock rate,
local clock source. When external clock input such as S/PDIF is
connected, the registers show the detection and external clock rate.
Additionally, the registers show the mode of optical digital input
interface. Although, a tester with Digi 003 rack reports this makes no
sense. Further investigation is required for Digi 003 series.
Besides, in Digi 002 rack, the S/PDIF format must be IEC 60958-4,
so-called professional.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a functionality to manage streaming.
The streaming is not controlled by CMP in IEC 61883-6. It's controlled by
IEEE 1394 write transaction to certain addresses.
Several clock sources are available, while there're no differences about
packet transmission. The value of SYT field in transmitted packets is
always zero. Thus, streams in both direction don't build synchronization.
And the device always requires received packets to transmit packets. This
driver keeps to transfer outgoing stream even if they're not required.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Digi 002/003 family uses its own format for data blocks. The format is
quite similar to AM824 in IEC 61883-6, while there're some differences:
* The Valid Bit Length (VBL) code is always 0x40 in Multi-bit Linear Audio
(MBLA) data channel.
* The first data channel includes MIDI messages, against IEC 61883-6
recommendation.
* The Counter field is always zero in MIDI conformant data channel.
* Sequence multiplexing in IEC 61883-6 is not applied to the MIDI
conformant data channel.
* PCM samples are scrambled in received AMDTP packets. We call the way
as Double-Oh-Three (DOT). The algorithm was discovered by
Robin Gareus and Damien Zammit in 2012.
This commit adds data processing layer to satisfy these differences.
There's a quirk about transmission mode for received packets. When this
driver applies non-blocking mode to outgoing packets with isochronous
channel 2 or more, after 15 to 20 seconds since playbacking, any PCM
samples causes noisy sound on the device. With isochronous channel 0 or 1,
this doesn't occur. As long as I investigated, this quirk is not observed
when applying blocking mode to the received packets.
This driver applies blocking mode to outgoing packets, while non-blocking
mode to incoming packgets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds a new driver for Digidesign 002/003 family. This commit
just creates/removes card instance according to bus event. More functions
will be added in following commits.
Digidesign 002/003 family consists of:
* Agere FW802B for IEEE 1394 PHY layer
* PDI 1394L40 for IEEE 1394 LINK layer and IEC 61883 interface
* ALTERA ACEX EP1K50 for IEC 61883 layer and DSP controller
* ADSP-21065L for signal processing
[minor cleanup using skip_spaces() by tiwai]
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Another attempt at drm-misc for 4.4 ...
- better atomic helpers for runtime pm drivers
- atomic fbdev
- dp aux i2c STATUS_UPDATE handling (for short i2c replies from the sink)
- bunch of constify patches
- inital kerneldoc for vga switcheroo
- some vblank code cleanups from Ville and Thierry
- various polish all over
* tag 'topic/drm-misc-2015-09-25' of git://anongit.freedesktop.org/drm-intel: (57 commits)
drm/irq: Add drm_crtc_vblank_count_and_time()
drm/irq: Rename drm_crtc -> crtc
drm: drm_atomic_crtc_get_property should be static
drm/gma500: Remove DP_LINK_STATUS_SIZE redefinition
vga_switcheroo: Set active attribute to false for audio clients
drm/core: Preserve the fb id on close.
drm/core: Preserve the framebuffer after removing it.
drm: Use vblank timestamps to guesstimate how many vblanks were missed
drm: store_vblank() is never called with NULL timestamp
drm: Clean up drm_calc_vbltimestamp_from_scanoutpos() vbl_status
drm: Limit the number of .get_vblank_counter() retries
drm: Pass flags to drm_update_vblank_count()
drm/i915: Fix vblank count variable types
drm: Kill pixeldur_ns
drm: Stop using linedur_ns and pixeldur_ns for vblank timestamps
drm: Move timestamping constants into drm_vblank_crtc
drm/fbdev: Update legacy plane->fb refcounting for atomic restore
drm: fix kernel-doc warnings in drm_crtc.h
vga_switcheroo: Sort headers alphabetically
drm: Spell vga_switcheroo consistently
...
SND_SOC_DAIFMT_{IB|NB}_{IF|NF} are defined as inverting or not BCLK or
FRM relatively to what is standard for the specified DAI hardware audio
format. Consequently, the absolute polarities of these signals cannot be
derived only from these settings as this driver did. The format has to
be taken into account too.
This fixes inverted left/right channels in I²S mode.
Signed-off-by: Benoît Thébaudeau <benoit@wsystem.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_release_substream() always calls hw_free op when the stream
was opened. This is superfluous in most cases because it's been
already released via explicit hw_free ioctl. Although this double
call is usually OK as this callback should be written to be called
multiple times, it's better to avoid superfluous calls.
Reported-by: Vinod Koul <vinod.koul@intel.com>
Tested-by: Jeeja Kp <jeeja.kp@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit moves the codes related to data block processing from packet
streaming layer to AM824 layer.
Each driver initializes amdtp stream structure for AM824 data block by
calling amdtp_am824_init(). Then, a memory block is allocated for AM824
specific structure. This memory block is released by calling
amdtp_stream_destroy().
When setting streaming parameters, it calls amdtp_am824_set_parameters().
When starting packet streaming, it calls amdtp_stream_start(). When
stopping packet streaming, it calls amdtp_stream_stop().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit renames some macros just related to AM824 format. In later
commit, they're moved to AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Setting the format of PCM substream to AMDTP stream structure is important
to set a handler to copy actual PCM samples between buffers. The
processing should be in data block processing layer because essentially
it has no relationship to packet streaming.
This commit renames PCM format setting function to prepare for integrating
AM824 layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, MIDI messages are transferred in MIDI conformant data
channel. Essentially, packet streaming layer is not responsible for MIDI
functionality.
This commit moves MIDI trigger helper function from the layer to AM824
layer. The rest of codes related to MIDI functionality will be moved in
later commits.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, several types of data are available in AM824 format. The
data is transferred in each data channel. The position of data channel in
data block differs depending on model.
Current implementation has an array to map the index of data channel in an
data block to the position of actual data channel. The implementation
allows each driver to access the mapping directly.
In later commit, the mapping is in specific structure pushed into an
opaque pointer. Helper functions are required.
This commit adds the helper functions for this purpose. In IEC 61883-6,
AM824 format supports many data types, while this specification easily
causes over-engineering. Current AM824 implementation is allowed to handle
two types of data, Multi Bit Linear Audio data (=PCM samples) and MIDI
conformant data (=MIDI messages).
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, PCM frames are transferred in Multi Bit Linear Audio data
channel. The data channel transfers 16/20/24 bit PCM samples. Thus, PCM
substream has a constrain about it.
This commit moves codes related to the constraint from packet streaming
layer to AM824 data block processing layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The value of FDF field in CIP header is protocol-dependent. Thus, it's
better to allow data block processing layer to decide the value in any
timing.
In AM824 data format, the value of FDF field in CIP header indicates
N-flag and Nominal Sampling Frequency Code (sfc). The N-flag is for
switching 'Clock-based rate control mode' and 'Command-based rate control
mode'. In our implementation, 'Clock-based rate control mode' is just
supported. Therefore, When sampling transfer frequency is decided, then
the FDF can be set.
This commit replaces 'amdtp_stream_set_parameters' with
'amdtp_am824_set_parameters' to set the FDF. This is the same timing
to decide the ration between the number of data blocks and the number of
PCM frames.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This commit adds data block processing layer for AM824 format. The new
layer initializes streaming layer with its value for fmt field.
Currently, most implementation of data block processing still remains
streaming layer. In later commits, these codes will be moved to the layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, data block processing layer will be newly added. This
layer will be named as 'amdtp-am824'.
This commit renames current amdtp file to amdtp-stream, to distinguish it
from the new layer.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some vendor specific protocol uses its own value for fmt/fdf fields in
CIP header.
This commit support to set arbitrary values for the fields.
In IEC 61883-6, NO-DATA code is defined for FDF field. A packet with this
code includes no data even if it includes some data blocks. This commit
still leaves a condition to handle this special packet.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA PCM framework uses PCM buffer with a concept of 'period' to
synchronize userspace operations to hardware for nearly-realtime
processing. Each driver implements snd_pcm_period_elapsed() to tell across
of the period boundary to ALSA PCM middleware. To call the function, some
drivers utilize hardware interrupt handlers, the others count handled PCM
frames.
Drivers for sound units on IEEE 1394 bus are the latter. They use two
buffers; PCM buffer and DMA buffer for IEEE 1394 isochronous packet. PCM
frames are copied between these two buffers and 'amdtp_stream' structure
counts the handled PCM frames. Then, snd_pcm_period_elapsed() is called if
required.
Essentially, packet streaming layer should not be responsible for PCM
frame processing. The PCM frame processing should be handled in each data
block processing layer as a result of handling data blocks. Although, PCM
frame counting is a common work for all of protocols which ALSA firewire
stack is going to support.
This commit adds two new helper functions as interfaces between packet
streaming layer to data block processing layer. In future, each data block
processing layer implements these functions. The packet streaming layer
calls data block processing layer per packet by calling the functions. The
data block processing layer processes data blocks and PCM frames, and
returns the number of processed PCM frames. Then the packet streaming layer
calculates handled PCM frames and calls snd_pcm_period_elapsed().
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In future commit, interface between data block processing layer and packet
stream processing layer is defined. These two layers communicate the
number of data blocks and the number of PCM frames.
The data block processing layer has a responsibility for calculating the
number of PCM frames. Therefore, 'dual wire' of Dice quirk should be
handled in data block processing layer.
This commit adds a member of 'frame_multiplier'. This member represents
the ratio of the number of PCM frames against the number of data blocks.
Usually, the value of this member is 1, while it's 2 in Dice's 'dual wire'.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In IEC 61883-6, one data block represents one event. In ALSA, the event is
one PCM frame. Therefore, when processing one data block, current
implementation counts one PCM frame.
On the other hand, Dice platform has a quirk called as 'dual wire' at
higher sampling rate. In detail, see comment of commit 6eb6c81eee
("ALSA: dice: Split stream functionality into a file").
Currently, to handle this quirk, AMDTP stream structure has a
'double_pcm_frames' member. When this is enabled, two PCM frames are
counted. Each driver set this flag by accessing the structure member
directly.
In future commit, some members related to AM824 data block will be moved
to specific structure, to separate packet streaming layer and data block
processing layer. The access will be limited by opaque pointer.
For this reason, this commit adds an argument into
amdtp_stream_set_parameter() to set the flag.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently, amdtp_stream_set_parameters() returns no error even if wrong
arguments are given. This is not good for streaming layer because drivers
can continue processing ignoring capability of streaming layer.
This commit changes this function to return error code.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In later commit, some members related to AM824 data format will be moved
from AMDTP stream structure to data block structure. This commit is a
preparation for it. Additionally, current layout of AMDTP stream structure
is a bit mess by several extensions. This commit also arranges the layout.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We want to verify that "value" is either zero or one, so we test if it
is greater than one. Unfortunately, this is a signed int so it could
also be negative. I think this is harmless but it introduces a static
checker warning. Let's make "value" unsigned.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit b4508d0f95 ("ASoC: db1200: Use static DAI format setup") switched
the db1200 driver over to using static DAI format setup instead of a
callback function. But the commit only added the dai_fmt field to one of
the three DAI links in the driver. This breaks audio on db1300 and db1550.
Add the two missing dai_fmt settings to fix the issue.
Fixes: b4508d0f95 ("ASoC: db1200: Use static DAI format setup")
Reported-by: Manuel Lauss <manuel.lauss@gmail.com>
Tested-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
For display audio, call the sync_audio_rate callback function
to do the synchronization between gfx driver and audio driver.
Signed-off-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
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Merge tag 'asoc-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.4
A first batch of updates targetted at v4.4. There are no substantial
core fixes here, the biggest block of changes is updates to the rcar
drivers and the addition of a CODEC driver for the AK4613.
Lenovo Thinkpads with recent Realtek codecs seem suffering from click
noises at power transition since the introduction of widget power
saving in 4.1 kernel. Although this might be solved by some delays in
appropriate points, as a quick workaround, just disable the
power_save_node feature for now. The gain it gives is relatively
small, and this makes the situation back to pre 4.1 time.
This patch ended up with a bit more code changes than usual because
the existing fixup for Thinkpads is highly chained. Instead of adding
yet another chain, combine a few of them into a single fixup entry, as
a gratis cleanup.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
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Merge tag 'asoc-fix-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v4.3
A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers. Nothing especially
stands out, it's mostly all device specific and relatively minor.
The active attribute in struct vga_switcheroo_client denotes whether
the outputs are currently switched to this client. The attribute is
only meaningful for vga clients. It is never used for audio clients.
The function vga_switcheroo_register_audio_client() misuses this
attribute to store whether the audio device is fully initialized.
Most likely there was a misunderstanding about the meaning of
"active" when this was added.
Comment from Takashi's review:
"Not really. The full initialization of audio was meant that the audio
is active indeed. Admittedly, though, the active flag for each audio
client doesn't play any role because the audio always follows the gfx
state changes, and the value passed there doesn't reflect the actual
state due to the later change. So, I agree with the removal of the
flag itself -- or let the audio active flag following the
corresponding gfx flag. The latter will make the proc output more
consistent while the former is certainly more reduction of code."
Set the active attribute to false for audio clients. Remove the
active parameter from vga_switcheroo_register_audio_client() and
its sole caller, hda_intel.c:register_vga_switcheroo().
vga_switcheroo_register_audio_client() was introduced by 3e9e63dbd3
("vga_switcheroo: Add the support for audio clients"). Its use in
hda_intel.c was introduced by a82d51ed24 ("ALSA: hda - Support
VGA-switcheroo").
v1.1: The changes above imply that in find_active_client() the call
to client_is_vga() is now superfluous. Drop it.
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Lukas Wunner <lukas@wunner.de>
[danvet: Add Takashi's clarification to the commit message.]
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Vetter <daniel.vetter@ffwll.ch>
Also move the include of sound/hda_verbs.h to rl6347a.h because it is used
in rl6347a.h.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The Tegra HD-audio controller driver causes deadlocks when loaded as a
module since the driver invokes request_module() at binding with the
codec driver. This patch works around it by deferring the probe in a
work like Intel HD-audio controller driver does. Although hovering
the codec probe stuff into udev would be a better solution, it may
cause other regressions, so let's try this band-aid fix until the more
proper solution gets landed.
Reported-by: Thierry Reding <treding@nvidia.com>
Tested-by: Thierry Reding <treding@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As far as I can see, having an invalid ->tstamp_mode is harmless, but
adding a check silences a static checker warning.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mackie Onyx Satellite has two usage; standalone and with base station.
These two modes has different stream formats. In standalone mode, rx data
block includes 2 Multi Bit Linear Audio (MBLA) data channels and tx data
block includes 2. With base station, they're 6 and 2.
Although, with base station, the actual tx packet include wrong value in
'dbs' field in its CIP header. This quirk causes packet streaming layer to
detect packet discontinuity and to stop PCM substream.
This is a response of 'single' subfunction 'extended stream format
information' command in AV/C Stream Format Information Specification 1.1.
It means that a data block in tx packets includes 2 MBLA data channels.
$ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffffffff
response: 000: 0c ff bf c0 01 00 00 00 ff 00 90 40 03 02 01 02
response: 010: 06
Current OXFW driver parses the response and detects stream formats
correctly.
$ cat /proc/asound/card1/firewire/formation
...
Output Stream from device:
Rate PCM MIDI
* 48000 2 0
44100 2 0
88200 2 0
96000 2 0
On the other hand, in actual tx CIP, the 'dbs' field has 6. But the number
of quadlets in CIP payload is not multiple of 6. Seeing the subtraction of
two successive payload quadlets, it should be multiple of 2.
payload CIP CIP
quadlets header0 header1
24 00060052 9002ffff
24 0006005e 9002ffff
26 0006006a 9002ffff
24 00060077 9002ffff
24 00060083 9002ffff
26 0006008f 9002ffff
24 0006009c 9002ffff
24 000600a8 9002ffff
26 000600b4 9002ffff
24 000600c1 9002ffff
This commit adds support for this quirk to OXFW driver, by using
CIP_WRONG_DBS flag. When this flag is set, packet streaming layer uses
the value of its 'data_block_quadlets' member instead of the value in CIP
header. This value is already set by OXFW driver and no discontinuity is
detected.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
PCM receive and transmit DMA requestor lines were reverted, breaking the
PCM playback interface for PXA platforms using the sound/soc/ variant
instead of the sound/arm variant.
The commit below shows the inversion in the requestor lines.
Fixes: d65a14587a ("ASoC: pxa: use snd_dmaengine_dai_dma_data")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Broadwell can not triger the IRQ falling and rising simultaneously,
so it can not detect jack-in and jack-out simultaneously.
We add a flag "jd_invert" to platform data. If this flag is set,
codec IRQ will be set to invert that forces IRQ as pulse when jack-in
and jack-out.
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add DMI data for Buddy project.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: John Lin <john.lin@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current code incorrectly treats dai format for AC97 as bit mask
whereas it's actually an integer value. This causes DAI formats
other than AC97 (e.g. DSP_B) to trigger AC97 related code,
which is incorrect and breaks functionality. This patch fixes
the code to correctly compare values to determine AC97 or not.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Acked-by: Timur Tabi <timur@tabi.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the capability to use multiple codecs on the same DAI linke where
one codec is used for playback and another one is used for capture.
Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS
microphone for capture.
No verification is made that the playback and capture codec setups are
compatible in terms of number of TDM slots (where applicable).
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The hdmi stub codec has not been used since refactoring of OMAP HDMI
audio support.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We need to create CPU DAI for each endpoint instance. For this we
should have one DMIC DAI, one HDA DAI and SSP DAI. Thus, DMIC23,
HDA-SPK/AMIC was not required so this patch removes them
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the error path so that we can free the allocated memory on the error
path instead of releasing them individually on each error.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We have requested for the firmware but we have missed releasing it both
on success and on error path.
While checking the code it turned out that the requested firmware is not
even used. More over the same firmware is being loaded by
wm0010_stage2_load().
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
the max register value of mic boost pga should be 3.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
wm8962 can't support 64k sample rate. When playing a 64KHz wave file,
'Unsupported rate 64000Hz' will be prompted.
Signed-off-by: Zidan Wang <zidan.wang@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Make SND_SOC_ROCKCHIP_MAX98090 and SND_SOC_ROCKCHIP_RT5645 depend on
CLKDEV_LOOKUP to fix below build warning:
warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects
SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K &&
!UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP)
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The '\n' at the end of the format string is not needed. It adds an extra
line break when doing
cat /proc/interrupts
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it.
Remove .owner field if calls are used which set it automatically
Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci
CC: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The slot_width is for essentially same thing. Instead of storing
bclk_lrclk_ratio, just store the slot_width. Comments has been updated
accordingly and some variable names changed to more descriptive.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Apply PTR_ERR to the value that was recently assigned.
The semantic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression x,y;
@@
if (IS_ERR(x) || ...) {
... when any
when != IS_ERR(...)
(
PTR_ERR(x)
|
* PTR_ERR(y)
)
... when any
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
Store return value of of_get_property() to a pointer of __be32 type as
device tree has big endian type. This fixes a sparse warning couple of
lines later when be32_to_cpup() is used to convert from big endian to
cpu endian.
The whole conversion is not really necessary, as we are only checking
if the value is zero or not, but I wanted to add it to remind in the
future that the data has to be converted before use. Compiler should
optimize the unnecessary operations away.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Earlier revisions of the wm5110/8280 silicon require a slightly more
complex procedure to enable analogue inputs. This patch adds this into
the driver.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We will occasionally require to take different action based on if an
input is analog or digital so add a helper function to return if an
input is analog.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the addition of the WILL_PMU / WILL_PMD several of the switches in
arizona.c no longer cover all cases or have a default case. Whilst this
isn't causing any problems in the interests of robustness add default
cases.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The previous fix of pxa library support, which was introduced to fix the
library dependency, broke the previous SoC behavior, where a machine
code binding pxa2xx-ac97 with a coded relied on :
- sound/soc/pxa/pxa2xx-ac97.c
- sound/soc/codecs/XXX.c
For example, the mioa701_wm9713.c machine code is currently broken. The
"select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for
compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared
in sound/arm/Kconfig and sound/soc/pxa/Kconfig.
Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct
pxa2xx-ac97 compilation.
Fixes: 846172dfe3 ("ASoC: fix SND_PXA2XX_LIB Kconfig warning")
Signed-off-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@kernel.org>
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") added AUDIO_CLKOUT
support for ADG. But single AUDIO_CLKOUT needs clkout_name[CLKOUT],
not clkout_name[i]. Kernel will have NULL pointer access without this
patch.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
71a0138ab("ASoC: ak4642: enable to use MCKO as fixed rate output
pin on DT") added new FS() macro, but x86 already has it in
arch/x86/include/uapi/asm/ptrace-abi.h
This patch exchange FS() to FSs() to avoid redefinition warning
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") uses
of_clk_add_provider() which is requesting struct clk_onecell_data.
But it is COMMON_CLK feature. SND_SOC_RCAR depends on COMMON_CLK
This patch also solved compile error of 7486d80f7("ASoC: rsnd: remove
unneeded sh_clk header")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The sun4i, sun5i and sun7i SoC families have a built-in codec, capable
of both audio capture and playback.
While this is called a codec by Allwinner, it really is an in-SoC
combination of a codec and a DAI, with its own DAC/ADC and amplifiers
in a single memory-mapped controller.
The capture part has been left out for now, and will be added eventually.
Signed-off-by: Emilio López <emilio@elopez.com.ar>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The function get_current_pipe_id() was not being used.
Signed-off-by: Sudip Mukherjee <sudip@vectorindia.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Adds DT binding for explicitly choosing a tdm mask for DAI and uses it
in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been
changed.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Implements set_tdm_slot() callback for mcasp. Channel constraints are
updated according to the configured tdm mask and slots each time
set_tdm_slot() is called. The special case when slot width is set to
zero is allowed and it means that slot width is the same as the sample
width.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Before this patch the set_tdm_slots() callback did not store the value
of slot width anywhere. The tdm support only worked if selected slot
width was equal to the sample width. With this patch all sample widths
that fit into the slot width are supported. There unused bits are
filled unnecessarily in the capture direction, but the other end of
the i2s bus should be able to ignore them.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
devm_snd_dmaengine_pcm_register() is guarded by
CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM.
Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has AUDIO_CLKOUT (in Gen1/Gen2) AUDIO_CLKOUT1/2/3 (in Gen3)
This patch support these patches as clock provider.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
undefined clock is not error. Accept such case. And this is prepare
for clock out support in the same time.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
It didn't have "\n", and indicated different data
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Current ADG clock calculation needs ADG and SSI settings.
Thus, SSI side clock request function depends on ADG settings.
After reconsideration, we can close this method inside ADG.
This function uses new method. And it becomes preparation for
AUDIO_CLKOUT support.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ADG is special IP since it doesn't have MSTP. And now, ADG has common
mod base register access. We can now setup ADG initial setting when
probe timing. It is needed if sound card is based on AUIDO_CLK which
is used as Master clock.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has SSI/SRC/DVC/MIX/ADG modules, and these have original
register mapping. Thus this driver is using regmap field, and each module
is using it based on each module ID.
Sometimes, each module needs other module to controlling. but current each
function is using just "mod" as parameter name. This is confusable.
For example, if SSI0 and SRC2 are connected, and if SRC module function
has bug of module access, and if it needs to control connected SSI,
SRC function will access to SSI2 (It should access to SSI0, but it uses
SRC's ID 2). This is easy to happen in current driver style.
To avoid this kind of confusable trouble, this patch adds module confirm
macro for debug purpose.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound has ADG IP, but it is special device.
(It is clock generater, and it doesn't need MSTP)
Thus, ADG didn't use mod base common method on rsnd driver.
But it can be confusable point. Let's use common method
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound driver has SSI/SRC/DVC/CTU/MIX, and these are controlled
as modules. And these module are member of each modules's private data.
It used own method to get module pointer, but Let's use common method
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Renesas sound Gen3 is updated version of Gen2. We need to update
driver for it, but basically it should works as Gen2 compatible.
This is initial support for Gen3. Gen3 specific feature will be
incrementally added in the future
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
sh_clk header is not needed, and it will create confusion.
Let's remove it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_src_pcm_new() is used only from Gen2. make it clear in function name,
and remove unneeded Gen1 check.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
rsnd_dma_to_xxx() macro should exist in same place
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
ak4642 chip can output clock via MCKO pin as audio reference clock.
This patch supports it on DT
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The TX and RX direction share the same bit clock and frame sync, so
the samplerate must be the same to both directions.
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
These platform drivers have a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This platform driver has a OF device ID table but the OF module
alias information is not created so module autoloading won't work.
Signed-off-by: Luis de Bethencourt <luis@debethencourt.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
