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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ALSA: firewire-lib: add helper functions as interfaces between packet streaming layer and data block processing layer

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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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 6eb6c81e
("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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi 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: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In PCM core, when hw_params() in each driver returns error, the state of
PCM substream is kept as 'open'. In this case, current drivers for sound
units on IEEE 1394 bus doesn't decrement substream counter in hw_free()
correctly. This causes these drivers to keep streams even if not
required.

This commit fixes this bug. When snd_pcm_lib_alloc_vmalloc_buffer()
fails, hw_params function in each driver returns without incrementing the
counter.
Reported-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The patch was generated using fixed coccinelle semantic patch
scripts/coccinelle/api/memdup.cocci [1].

[1]: http://permalink.gmane.org/gmane.linux.kernel/2014320Signed-off-by: NAndrzej Hajda <a.hajda@samsung.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface.
This chip includes ARM7 core, and loads and runs program. The firmware
is stored in on-board memory and loaded every powering-on from it.

Echo Audio ships several versions of firmwares for each model. These
firmwares have each quirk and the quirk changes a sequence of packets.

As long as I investigated, AudioFire2/AudioFire4/AudioFirePre8 have a
quirk to transfer a first packet with 0x02 in its dbc field. This causes
ALSA Fireworks driver to detect discontinuity. In this case, firmware
version 5.7.0, 5.7.3 and 5.8.0 are used.

Payload  CIP      CIP
quadlets header1  header2
02       00050002 90ffffff <-
42       0005000a 90013000
42       00050012 90014400
42       0005001a 90015800
02       0005001a 90ffffff
42       00050022 90019000
42       0005002a 9001a400
42       00050032 9001b800
02       00050032 90ffffff
42       0005003a 9001d000
42       00050042 9001e400
42       0005004a 9001f800
02       0005004a 90ffffff
(AudioFire2 with firmware version 5.7.)

$ dmesg
snd-fireworks fw1.0: Detect discontinuity of CIP: 00 02

These models, AudioFire8 (since Jul 2009 ) and Gibson Robot Interface
Pack series uses the same ARM binary as their firmware. Thus, this
quirk may be observed among them.

This commit adds a new member for AMDTP structure. This member represents
the value of dbc field in a first AMDTP packet. Drivers can set it with
a preferred value according to model's quirk.
Tested-by: NJohannes Oertei <johannes.oertel@uni-due.de>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This reverts commit 9c6893e0.

The fix is superseded by the next commit as a better implementation
for supporting AudioFire2/AudioFire4/AudioFirePre8 quirks.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fireworks uses TSB43CB43(IceLynx-Micro) as its IEC 61883-1/6 interface.
This chip includes ARM7 core, and loads and runs program. The firmware
is stored in on-board memory and loaded every powering-on.

Echo Audio ships several versions of firmwares for each model. These
firmwares have each quirk and the quirk changes a sequence of packets.

AudioFire2 has a quirk to transfer a first packet with non-zero in
its dbc field. This causes ALSA Fireworks driver to detect discontinuity.
As long as I investigated, firmware 5.7, 5.7.6 and 5.8 have this quirk.

This commit adds a support for the quirk to handle AudioFire2 packets.
For safe, CIP_SKIP_INIT_DBC_CHECK is applied to all versions of
AudioFire2's firmwares.

02 00050002 90ffffff <-
42 0005000a 90013000
42 00050012 90014400
42 0005001a 90015800
02 0005001a 90ffffff
42 00050022 90019000
42 0005002a 9001a400
42 00050032 9001b800
02 00050032 90ffffff
42 0005003a 9001d000
42 00050042 9001e400
42 0005004a 9001f800
02 0005004a 90ffffff
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

They're based on DM1500 (ArchWave produced), and BeBoB version 3 is
installed.

$ cat /proc/asound/FCA610/firewire/firmware
Manufacturer:	bridgeCo
Protocol Ver:	3
Build Ver:		0
GUID:		0x001564000002AD73
Model ID:		0x03
Model Rev:		0
Firmware Date:	20121102
Firmware Time:	153431
Firmware ID:	0x610
Firmware Ver:	8348
Base Addr:		0x400C0080
Max Size:		1422624
Loader Date:	20121015
Loader Time:	104710
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Behringer FCA610 transmits packets with periodic noisy PCM samples
when receiving no streams, and generates a bit noisy sound.

ALSA BeBoB driver is programmed to establish both in/out connections
when starting streaming, then transfers packets as userspace applications
requested. This means that there's a case that one of incoming/outgoing
streams is running, to save CPU and bandwidth usage. Although, it's natural
to start transferring packets in both direction.

This commit makes this driver to keeps duplex streams always.
Tested-by: NKim Tore Jensen <kim@incendio.no>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

PrismSound Orpheus, Behringer UFX1604 and FCA610 work with BeBoB v3, and
they're confirmed to transmit discontinuous packets in the beginning of
streaming.

payload    CIP headers
  8        0x00070000 0x9002FFFF
  8        0x00070000 0x9002FFFF
  8        0x00070000 0x9002FFFF
  8        0x00070008 0x9002FFFF <-
  8        0x00070008 0x9002FFFF
  8        0x00070008 0x9002FFFF
  8        0x00070008 0x9002FFFF
  8        0x00070008 0x9002FFFF
  8        0x00070008 0x9002FFFF
232        0x00070000 0x9002E798 <-
232        0x00070008 0x9002FB99
232        0x00070010 0x90021398
  8        0x00070018 0x9002FFFF
(This sample was got with Behringer FCA610 and FFADO library.)

This commit sets CIP_EMPTY_HAS_WRONG_DBC and CIP_SKIP_DBC_ZERO_CHECK to
ignore these discontinuities.
Tested-by: NKim Tore Jensen <kim@incendio.no>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Behringer FCA610 and UFX1604 is confirmed to require more time till
transmitting packets after establishing connections. This seems to
be a quirk of DM1500 ASIC which ArchWave produced.

For this quirk, this commit extends the time to wait up to 2 seconds.
As a result, in worst cases, below userspace functions require 2 seconds
to return.
 - snd_pcm_prepare()
 - snd_pcm_hw_params()
 - snd_pcm_recover()
Tested-by: NKim Tore Jensen <kim@incendio.no>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

BeBoB installed devices have BeBoB register area. This area stores
basic information about its firmware. A register has its protocol
version.

This commit adds 'version' member and store the device's protocol
version to handle v3 quirks in following commits.
Tested-by: NKim Tore Jensen <kim@incendio.no>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In previous commits, this driver can detect the source of clock as mush
as possible. SYT-Match mode is also available.

This commit purge the restriction.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The old string literals were completely replaced by new normalized
representation.

This commit obsoletes it.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This commit changes function prototype and its processing. As a result,
function caller can execute additional processing according to detected
clock source.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ALSA: bebob: preparation for replacing string literals by normalized representation for model-dependent structures

Previous commit adds a enumerator as a normalized representation of
clock source, while model-dependent structures still use string literals
for this purpose.

This commit is a preparation for replacement.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Previous commit allows this driver to detect several types of clock
source, while there's no normalized expression for it.

This commit adds a new enumerator for this purpose.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

With BeBoB version 3, current ALSA BeBoB driver detects the type of
current clock signal source wrongly. This is due to a lack of proper
implementation to parse the information.

This commit renews the parser. As a result, this driver detects
SYT-Match clock signal, thus it can start streams with two modes;
SYT-Match mode and the others. SYT-Match mode will be supported in future
commits.

There's a constrain about detected internal/external clock source.
When detecting external clock source, this driver allows userspace
applications to use current sampling rate only. This is due to consider
abour synchronization to external clock sources such as S/PDIF, ADAT or
word-clock.

According to several information from some devices, I guesss that the
internal clock of most devices synchronize to IEEE 1394 cycle start
packet. In this case, by a usual way, it's detect as 'Sync type
of output Music Sub-Unit' connected to 'Sync type of PCR output Unit
(oPCR)', and this driver judges it as internal clock. Therefore,
userspace applications is allowed to request arbitrary supported sampling
rates.

On the other hand, several devices based on BeBoB version 3 have
additional internal clock. In this case, by a usual way, it's detect as
'Sync/Additional type of External input Unit'. Unfortunately, there's no
way to distinguish this sync type from the other external clock sources
such as word-clock. In this case, this driver handles it as external and
userspace applications is forced to use current sampling rate.

I note that when the source of clock is detected as 'Isochronous stream
type of input PCR[0]', it's under 'SYT-Match' mode. In this mode, the
synchronization clock is generated according to SYT-series in received
packets. In this case, this driver generates the series by myself. I
experienced this mode often make the device silent suddenly during
playbacking. This means that the mode is easy to lost synchronization.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When detecting packet discontinuity, handle_in_packet() returns minus value
and this value is assigned to unsigned int variable, then the variable has
huge value. As a result, the variable causes buffer-over-run in
handle_out_packet(). This brings invalid page request and system hangup.

This commit fixes the bug to add a new argument into handle_in_packet()
and the number of handled data blocks is assignd to it. The function
return value is just used to check error.

I also considered to change the type of local variable to 'int' in
in_stream_callback(). This idea is based on type-conversion in C standard,
while it may cause future problems when adding more works. Thus, I dropped
this idea.

Fixes: 6fc6b9ce('ALSA: firewire-lib: pass the number of data blocks in incoming packets to outgoing packets')
Reported-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This device is based on DM1000E, and BeBoB version 1 firmware is
installed.

$ cat /proc/asound/cards
 0 [Pro            ]: BeBoB - Mbox 2 Pro
                      DIGIDESIGN Mbox 2 Pro (id:1, rev:1),
		      GUID 00a07e0100a90000 at fw1.0, S400

$ cat /proc/asound/Pro/firewire/firmware
Manufacturer:	bridgeCo
Protocol Ver:	1
Build Ver:		0
GUID:		0x00A07E0100A90000
Model ID:		0x01
Model Rev:		1
Firmware Date:	20071031
Firmware Time:	034402
Firmware ID:	0xA9
Firmware Ver:	16777215
Base Addr:		0x20080000
Max Size:		1572864
Loader Date:	20051207
Loader Time:	205554

With this patch, ALSA BeBoB driver can start packet streaming to/from
this model, while as a default, internal multiplexer of this model is
not initialized and generates no sound even if the driver transfers
any packets with PCM samples. To hear any sounds from this model,
userspace applications should be developed to set parameters to the
internal multiplexer. You can see raw information in FFADO website:
http://subversion.ffado.org/wiki/AvcModels/DigiDesignMboxPro2Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When detecting zero in 'dbs' field of CIP header, this packet streaming
should be aborted because of avoiding division-by-zero. This is an error
in an aspect of IEC 61883-1, thus protocol error.

This commit use EPROTO instead of EIO. Actually, the returned value is
not used for userspace and this commit has no effect.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When detecting invalid value in 'dbs' field of CIP header or packet
discontinuity, current implementation reports the status by err_info().
In most cases this state is caused by model-specific issue due to
vendor's customization and should be reported to developers.

This commit use dev_err() instead of dev_info() for this purpose.
In the cases, packet streaming is aborted, thus no message floading
occurs.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some macros include my misunderstanding for IEC 61883-1 or -6.
Additionally, some fixed values appear on codes.

This commit replaces these with macros with proper names.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The naming rule for local functions was inconsistent. This commit
rename them with a consistent manner.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Former patches allow non-blocking streams to synchronize with timestamp.
This patch removes the restriction.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In previous commit, error handling for incoming packet processing is
outside of packetization. This is nice for reading the codes.

This commit applies this idea for outgoing packet processing, too.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Current implementation reuses the value of syt field in incoming packet to
outgoing packet for full duplex timestamp synchronization, while the number
of data blocks in outgoing packets refers to hard-coded table and the
synchronization cannot be applied to non-blocking stream.

This commit passes the number of data blocks from incoming packet
processing to outgoing packet processing for the synchronization. For
normal mode, isochronous callback handler is changed to generate the values
of syt and data blocks.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This function is called according to conditions between the value of
syt and streaming mode(blocking or non-blocking).

To simplify caller's work, this commit push these conditions to the
function.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In IEC 61883-6, the number of data blocks in a packet is limited up to
the value of SYT_INTERVAL. Current implementation is compliant to the
limitation, while it can cause buffer-over-run when the value of dbs
field in received packet is illegally large.

This commit adds a validator to detect such illegal packets to prevent
the buffer-over-run. Actually, the buffer is aligned to the size of memory
 page, thus this issue hardly causes system errors due to the room to page
alignment, as long as a few packets includes such jumbo payload; i.e.
a packet to several received packets.

Here, Behringer F-Control Audio 202 (based on OXFW 960) has a quirk to
postpone transferring isochronous packet till finish handling any
asynchronous packets. In this case, this model is lazy, transfers no
packets according to several cycle-start packets. After finishing, this
model pushes required data in next isochronous packet. As a result, the
packet include more data blocks than IEC 61883-6 defines.

To continue to support this model, this commit adds a new flag to extend
the length of calculated payload. This flag allows the size of payload
5 times as large as IEC 61883-6 defines. As a result, packets from this
model passed the validator successfully.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Some M-Audio devices require to receive bootup command just after
powering on, while codes in BeBoB driver doesn't work properly in
big-endian machine because the command should be aligned by
little-endian.

This commit fixes this bug. This fix should go to stable kernel.

Cc: Takayuki Shiroma <t.shiroma.oki@gmail.com>
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>