The ADCCONTROL4 and DACCONTROL1 registers are similar but not identical,
with the DACCONTROL1 having each field starting one bit higher than
ADCCONTROL4.
Instead of introducing a magic shift, add new constants for the values
in ADCCONTROL4 and use a second variable to setup the ADC.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This ensures that the clock is setup after its frequency has been set;
the existing code in set_dai_fmt may be called before the clock rate has
been set resulting in an incorrect configuration.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a
chmap has been already assigned, it returns as an error due to the
conflicting snd_ctl_add() result. However, this also clears the
already assigned chmap_kctl field via pcm_chmap_ctl_private_free(),
and becomes inconsistent in the later operation.
This patch adds the check of the conflicting chmap kctl before
actually trying to allocate / assign. The check failure is treated as
a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is
basically a driver bug and having the stack trace would help
developers to figure out the bad code path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In former commit, drivers in ALSA firewire stack always starts IT context
before IR context. If IR context starts after packets are transmitted by
peer unit, packet discontinuity may be detected because the context starts
in the middle of packet streaming. This situation is rare because IT
context usually starts immediately. However, it's better to solve this
issue. This is suppressed with CIP_SKIP_INIT_DBC_CHECK flag.
This commit enables the same feature as CIP_SKIP_INIT_DBC_CHECK.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, this module has no need to reuse parameters of
incoming packets for outgoing packets anymore. This commit arranges some
needless codes for outgoing packet processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In previous commit, this module has no need to reuse parameters of
incoming packets for outgoing packets anymore. This commit arranges some
needless codes for incoming packet processing.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In clause 6.3 of IEC 61883-6:2000, there's an explanation about processing
of presentation timestamp. In the clause, we can see "If a function block
receives a CIP, processes it and subsequently re-transmits it, then the
SYT of the outgoing CIP shall be the sum of the incoming SYT and the
processing delay." ALSA firewire stack has an implementation to partly
satisfy this specification. Developers assumed the stack to perform as an
Audio function block[1].
Following to the assumption, current implementation of ALSA firewire stack
use one software interrupt context to handle both of in/out packets. In
most case, this is processed in 1394 OHCI IR context independently of the
opposite context. Thus, this implementation uses longer CPU time in the
software interrupt context. This is not better for whole system.
Against the assumption, I confirmed that each ASIC for IEC 61883-1/6
doesn't necessarily expect it to the stack. Thus, current implementation
of ALSA firewire stack includes over-engineering.
This commit purges the implementation. As a result, packets of one
direction are handled in one software interrupt context and spends
minimum CPU time.
[1] [alsa-devel] [PATCH 0/8] [RFC] new driver for Echo Audio's Fireworks based devices
http://mailman.alsa-project.org/pipermail/alsa-devel/2013-June/062660.html
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In packet streaming protocol applied to TASCAM FireWire series, the value
of SYT field in CIP header is always zero, therefore it has no meaning.
There's no need to synchronize packets in both direction for the series.
In current implementation of ALSA firewire stack, driver for the series
uses incoming packet parameter for outgoing packet parameter to calculate
the number of data blocks. This can be simplified because the task of
corresponding driver is to transfer data blocks enough to sampling transfer
frequency.
This commit purges support of full duplex synchronization to prevent
over-engineering implementation.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Fireworks board module of Echo Audio, TSB43Cx43A (IceLynx Micro, iCEM)
is used to process payload of isochronous packets. There's an public
document of this chip[1]. This document is for firmware programmers to
transfer/receive AMDTP with IEC60958 data format, however in clause 2.5,
2.6 and 2.7, we can see system design to utilize the sequence of value in
SYT field of CIP header. In clause 2.3, we can see the specification of
Audio Master Clock (MCLK) from iCEM.
Well, this clock is actually not used for sampling clock. This can be
confirmed when corresponding driver transfer random value as the sequence
of SYT field. Even if in this case, the unit generates proper sound.
Additionally, in unique command set for this board module, the format
of CIP is changed; for IEC 61883-6 mode which we use, and for Windows
Operating System. In the latter mode, the whole 32 bit field in second CIP
header from Windows driver is used to represent counter of packets (NO-DATA
code is still used for packets without data blocks). If the master clock
was physically used by DSP on the board module, the Windows driver must
have transferred correct sequence of SYT field.
Furthermore, as long as seeing capacities of AudioFire2, AudioFire4,
AudioFire8, AudioFirePre8 and AudioFire12, these models don't support
SYT-Match clock source.
Summary, we have no need to relate incoming/outgoing packets. This commit
drops reusing SYT sequence of incoming packets for outgoing packets.
[1] Using TSB43Cx43A: S/PDIF over 1394 (2003, Texus Instruments
Incorporated)
http://www.ti.com/analog/docs/litabsmultiplefilelist.tsp?literatureNumber=slla148&docCategoryId=1&familyId=361
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Windows driver for BeBoB-based models mostly wait for transmitted packets,
then transfer packets to the models. This looks for the relationship
between incoming packets and outgoing packets to synchronize the sequence
of presentation timestamp.
However, the sequence between packets of both direction has no
relationship. Even if receiving NO-DATA packets, the drivers transfer
packets with meaningful value in SYT field. Additionally, the order of
starting packets is always the same, independently of the source of clock.
The corresponding driver is expected as a generator of presentation
timestamp and these models can select it as a source of sampling clock.
This commit drops reusing SYT sequence from ALSA bebob driver. The driver
always transfer packets with presentation timestamp generated by ALSA
firewire stack, without re-using the sequence of value in SYT field in
incoming packets to outgoing packets.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HD-audio reconfig function got broken in the recent kernels,
typically resulting in a failure like:
snd_hda_intel 0000:00:1b.0: control 3:0:0:Playback Channel Map:0 is already present
This is because of the code restructuring to move the PCM and control
instantiation into the codec drive probe, by the commit [bcd96557bd0a:
ALSA: hda - Build PCMs and controls at codec driver probe]. Although
the commit above removed the calls of snd_hda_codec_build_pcms() and
*_build_controls() at the controller driver probe, the similar calls
in the reconfig were still left forgotten. This caused the
conflicting and duplicated PCMs and controls.
The fix is trivial: just remove these superfluous calls from
reconfig_codec().
Fixes: bcd96557bd0a ('ALSA: hda - Build PCMs and controls at codec driver probe')
Reported-by: Jochen Henneberg <jh@henneberg-systemdesign.com>
Cc: <stable@vger.kernel.org> # v4.1+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the McASP is used as clock master and the reference clock is AUXCLK we
can have additional level of divider. The BCLK divider is limited to
maximum 32, if the desired bclk can not be reached with this, the AUXCLK
divider also needs to be used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the return value to error_pmm instead of the BCLK div and handle the
divider configuration to McASP within the function when the set flag is
true.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Change the first parameter to struct davinci_mcasp* from
struct snd_soc_dai*
The function internally does not use or need the DAI information.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Instead of hardwired IDs add defines for the available dividers.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since the recent rewrite of HD-audio infrastructure,
CONFIG_SND_HDA_RECONFIG has a slightly different meaning. In the
earlier versions, it implicitly assumed only the usage via hwdep sysfs
entries. Meanwhile, in the recent version, this option is meant to
enable the reconfig code in HD-audio driver, which may be used by the
patch loader without hwdep interface.
This patch tries to clarify the usage pattern a bit better, hopefully
avoid the further confusion.
Reported-by: Jochen Henneberg <jh@henneberg-systemdesign.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A switch statement looks a bit cleaner than an if statement
spread over 3 lines, as such update this to a switch.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We can't return a negative error code from the poll callback the return
type is unsigned and is checked against the poll specific flags we need
to return POLLERR if we encounter an error.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
stream can't be NULL here as we have just taken the address of it, so no
need for the check.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have a function that returns the appropriate flags for the stream
direction, so we should use it.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We can't return a negative error code from the poll callback the return
type is unsigned and is checked against the poll specific flags we need
to return POLLERR if we encounter an error.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure that the user can not start multiple streams with the same
direction.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM
operations are going to happen. This is needed to get suspend/resume
working correctly when the audio is using simple-card.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When audio and music units have some quirks in their sequence of packet,
it's really hard for non-owners to identify the quirks. Although developers
need dumps for sequence of packets, it's difficult for users who have no
knowledges and no equipments for this purpose.
This commit adds tracepoints for this situation. When users encounter
the issue, they can dump a part of packet data via Linux tracing framework
as long as using drivers in ALSA firewire stack.
Additionally, tracepoints for outgoing packets will be our help to check
and debug packet processing of ALSA firewire stack.
This commit newly adds 'snd_firewire_lib' subsystem with 'in_packet' and
'out_packet' events. In the events, some attributes of packets and the
index of packet managed by this module are recorded per packet.
This is an usage:
$ trace-cmd record -e snd_firewire_lib:out_packet \
-e snd_firewire_lib:in_packet
/sys/kernel/tracing/events/snd_firewire_lib/out_packet/filter
/sys/kernel/tracing/events/snd_firewire_lib/in_packet/filter
Hit Ctrl^C to stop recording
^C
$ trace-cmd report trace.dat
...
23647.033934: in_packet: 01 4073 ffc0 ffc1 00 000f0040 9001b2d1 122 44
23647.033936: in_packet: 01 4074 ffc0 ffc1 00 000f0048 9001c83b 122 45
23647.033937: in_packet: 01 4075 ffc0 ffc1 00 000f0050 9001ffff 002 46
23647.033938: in_packet: 01 4076 ffc0 ffc1 00 000f0050 9001e1a6 122 47
23647.035426: out_packet: 01 4123 ffc1 ffc0 01 010f00d0 9001fb40 122 17
23647.035428: out_packet: 01 4124 ffc1 ffc0 01 010f00d8 9001ffff 002 18
23647.035429: out_packet: 01 4125 ffc1 ffc0 01 010f00d8 900114aa 122 19
23647.035430: out_packet: 01 4126 ffc1 ffc0 01 010f00e0 90012a15 122 20
(Here, some common fields are omitted so that a line to be within 80
characters.)
...
One line represent one packet. The legend for the last nine fields is:
- The second of cycle scheduled for the packet
- The count of cycle scheduled for the packet
- The ID of node as source (hex)
- Some devices transfer packets with invalid source node ID in their CIP
header.
- The ID of node as destination (hex)
- The value is not in CIP header of packets.
- The value of isochronous channel
- The first quadlet of CIP header (hex)
- The second quadlet of CIP header (hex)
- The number of included quadlets
- The index of packet in a buffer maintained by this module
This is an example to parse these lines from text file by Python3 script:
\#!/usr/bin/env python3
import sys
def parse_ts(second, cycle, syt):
offset = syt & 0xfff
syt >>= 12
if cycle & 0x0f > syt:
cycle += 0x10
cycle &= 0x1ff0
cycle |= syt
second += cycle // 8000
cycle %= 8000
# In CYCLE_TIMER of 1394 OHCI, second is represented in 8 bit.
second %= 128
return (second, cycle, offset)
def calc_ts(second, cycle, offset):
ts = offset
ts += cycle * 3072
# In DMA descriptor of 1394 OHCI, second is represented in 3 bit.
ts += (second % 8) * 8000 * 3072
return ts
def subtract_ts(minuend, subtrahend):
# In DMA descriptor of 1394 OHCI, second is represented in 3 bit.
if minuend < subtrahend:
minuend += 8 * 8000 * 3072
return minuend - subtrahend
if len(sys.argv) != 2:
print('At least, one argument is required for packet dump.')
sys.exit()
filename = sys.argv[1]
data = []
prev = 0
with open(filename, 'r') as f:
for line in f:
pos = line.find('packet:')
if pos < 0:
continue
pos += len('packet:')
line = line[pos:].strip()
fields = line.split(' ')
datum = []
datum.append(fields[8])
syt = int(fields[6][4:], 16)
# Empty packet in IEC 61883-1, or NODATA in IEC 61883-6
if syt == 0xffff:
data_blocks = 0
else:
payload_size = int(fields[7], 10)
data_block_size = int(fields[5][2:4], 16)
data_blocks = (payload_size - 2) / data_block_size
datum.append(data_blocks)
second = int(fields[0], 10)
cycle = int(fields[1], 10)
start = (second << 25) | (cycle << 12)
datum.append('0x{0:08x}'.format(start))
start = calc_ts(second, cycle, 0)
datum.append("0x" + fields[5])
datum.append("0x" + fields[6])
if syt == 0xffff:
second = 0
cycle = 0
tick = 0
else:
second, cycle, tick = parse_ts(second, cycle, syt)
ts = calc_ts(second, cycle, tick)
datum.append(start)
datum.append(ts)
if ts == 0:
datum.append(0)
datum.append(0)
else:
# Usual case, or a case over 8 seconds.
if ts > start or start > 7 * 8000 * 3072:
datum.append(subtract_ts(ts, start))
if ts > prev or start > 7 * 8000 * 3072:
gap = subtract_ts(ts, prev)
datum.append(gap)
else:
datum.append('backward')
else:
datum.append('invalid')
prev = ts
data.append(datum)
sys.exit()
The data variable includes array with these elements:
- The index of the packet
- The number of data blocks in the packet
- The value of cycle count (hex)
- The value of CIP header 1 (hex)
- The value of CIP header 2 (hex)
- The value of cycle count (tick)
- The value of calculated presentation timestamp (tick)
- The offset between the cycle count and presentation timestamp
- The elapsed ticks from the previous presentation timestamp
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In callback function of isochronous context, modules can queue packets to
indicated isochronous cycles. Although the cycle to queue a packet is
deterministic by calculation, this module doesn't implement the calculation
because it's useless for processing.
In future, the cycle count is going to be printed with the other parameters
for debugging. This commit is the preparation. The cycle count is computed
by cycle unit, and correctly arranged to corresponding packets. The
calculated count is used in later commit.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In callback function of isochronous context, u32 variable is passed for
cycle count. The value of this variable comes from DMA descriptors of 1394
Open Host Controller Interface (1394 OHCI). In the specification, DMA
descriptors transport lower 3 bits for second field and full cycle field in
16 bits field, therefore 16 bits of the u32 variable are available. The
value for second is modulo 8, and the value for cycle is modulo 8,000.
Currently, ALSA firewire-lib module don't use the value of the second
field, because the value is useless to calculate presentation timestamp in
IEC 61883-6. However, the value may be useful for debugging. In later
commit, it will be printed with the other parameters for debugging.
This commit makes this module to handle the whole cycle count including
second. The value is calculated by cycle unit. The existed code is already
written with ignoring the value of second, thus this commit causes no
issues.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For reducing the noise from the headset output on ASUS UX501VW,
call the existing fixup, alc_fixup_headset_mode_alc668(), additionally.
Thread: https://bbs.archlinux.org/viewtopic.php?id=209554
Signed-off-by: Kaho Ng <ngkaho1234@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the DMAs are not being quiesced properly, it may lead to
stability issues, so the recommendation is to wait till DMAs are
stopped.
After setting the stop bit of RIRB/CORB DMA, we should wait for
stop bit to be set.
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>
ebus is a member of extended device and was never initialized, so
do this at device creation.
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>
Support new codecs for ALC234/ALC274/ALC294.
This three codecs was the same IC.
But bonding is not the same.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"header->number" can be up to USHRT_MAX and it comes from the ioctl so
it needs to be capped.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The stack object “r1” has a total size of 32 bytes. Its field
“event” and “val” both contain 4 bytes padding. These 8 bytes
padding bytes are sent to user without being initialized.
Signed-off-by: Kangjie Lu <kjlu@gatech.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The stack object “r1” has a total size of 32 bytes. Its field
“event” and “val” both contain 4 bytes padding. These 8 bytes
padding bytes are sent to user without being initialized.
Signed-off-by: Kangjie Lu <kjlu@gatech.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The stack object “tread” has a total size of 32 bytes. Its field
“event” and “val” both contain 4 bytes padding. These 8 bytes
padding bytes are sent to user without being initialized.
Signed-off-by: Kangjie Lu <kjlu@gatech.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
These platform drivers are lacking MODULE_ALIAS so module autoloading
doesn't work. Tested on corgi and poodle with kernel 4.4.
Signed-off-by: Andrea Adami <andrea.adami@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
During skl_nhlt_init(), acpi obj pointer is allocated and never
freed and remap address is not unmapped.
To fix this we should release the ACPI obj and also unmap the
nhlt address during cleanup of driver.
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>
name and sname allocated in widget create are not freed when
creation is successful, so free them.
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 mx6ul the General Purpose Register 1 (GPR1) contains the following
bits for configuring the direction of the SAI MCLKs:
SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR
Introduce the "fsl,sai-mclk-direction-output" optional property to allow
configuring the SAI_MCLK outputs.
Tested on a imx6ul-evk board.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
MX6UL may need to configure the General Purpose Register 1 (GPR1), so
it is better to add a new compatible string to differentiate.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This adds "DLGS7219" ACPI id for the codec.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Tested-by: Sathyanarayana Nujella <sathyanarayana.nujella@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
We only polling jd status when rt5645->pdata.jd_invert is true.
However, it should be done at all time since there will be no
interrupt for jd if we press a headset button and remove the
headset at the same time.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit eba65d179c1149cf79e68608d452631f33d7f017.
This broke audio on Veyron Jerry Chromebooks and I now cannot reproduce
the problem I was trying to fix even with this commit reverted, so it
seems that this was completely the wrong thing to do.
Reported-by: Enric Balletbo Serra <eballetbo@gmail.com>
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This reverts commit 5938448b99275cba95167c3f9d39ca9225fdad38.
It turns out that the commit that made these variables unused is wrong
so we're about to revert it. Bring back the variables in prepration.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
If someone powers down the DSP core (through routing changes
say) whilst a compressed record is in progress we can end up
using a freed pointer to the buffer object. When a compressed
audio stream is triggered we attach it to a buffer on a physical
DSP. This patch adds a detach of the buffer from the stream when
the stream is freed or when the DSP is powered down which avoids
the situation where we use a buffer when it is no longer valid.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
