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>
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>
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>
New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.
This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.
Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check of cval->cached should be zero-based (including master channel).
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In theory, the device may get suspended even at runtime PM suspend.
Currently we don't save the mixer state for autopm, and it may bring
inconsistency.
This patch removes the special handling for autosuspend.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
After the recent fix of runtime PM for USB-audio driver, we got a
lockdep warning like:
=============================================
[ INFO: possible recursive locking detected ]
4.2.0-rc8+ #61 Not tainted
---------------------------------------------
pulseaudio/980 is trying to acquire lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
but task is already holding lock:
(&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
This comes from snd_usb_autoresume() invoking down_read() and it's
used in a nested way. Although it's basically safe, per se (as these
are read locks), it's better to reduce such spurious warnings.
The read lock is needed to guarantee the execution of "shutdown"
(cleanup at disconnection) task after all concurrent tasks are
finished. This can be implemented in another better way.
Also, the current check of chip->in_pm isn't good enough for
protecting the racy execution of multiple auto-resumes.
This patch rewrites the logic of snd_usb_autoresume() & co; namely,
- The recursive call of autopm is avoided by the new refcount,
chip->active. The chip->in_pm flag is removed accordingly.
- Instead of rwsem, another refcount, chip->usage_count, is introduced
for tracking the period to delay the shutdown procedure. At
the last clear of this refcount, wake_up() to the shutdown waiter is
called.
- The shutdown flag is replaced with shutdown atomic count; this is
for reducing the lock.
- Two new helpers are introduced to simplify the management of these
refcounts; snd_usb_lock_shutdown() increases the usage_count, checks
the shutdown state, and does autoresume. snd_usb_unlock_shutdown()
does the opposite. Most of mixer and other codes just need this,
and simply returns an error if it receives an error from lock.
Fixes: 9003ebb13f ('ALSA: usb-audio: Fix runtime PM unbalance')
Reported-and-tested-by: Alexnader Kuleshov <kuleshovmail@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the Gustard DAC-X20U.
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The input terminal parser recurses into the referenced clock entity to verify
it is existant and thus the terminal descriptor is valid. The actual property
values of the term instance which is initially parsed must not be overriden by
the recursion. For this to work the term properties have to be assigned after
recursing into the referenced clock entity descriptors.
Signed-off-by: Julian Scheel <julian@jusst.de>
Acked-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The fix for deadlock in PM in commit [1ee23fe07e: ALSA: usb-audio:
Fix deadlocks at resuming] introduced a new check of in_pm flag.
However, the brainless patch author evaluated it in a wrong way
(logical AND instead of logical OR), thus usb_autopm_get_interface()
is wrongly called at probing, leading to unbalance of runtime PM
refcount.
This patch fixes it by correcting the logic.
Reported-by: Hans Yang <hansy@nvidia.com>
Fixes: 1ee23fe07e ('ALSA: usb-audio: Fix deadlocks at resuming')
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The M-Audio Transit exposes an interface with a SYNC_NONE attribute.
This is not a valid value according to the USB audio classspec. However
there is a sync endpoint associated to this record. Changing the logic to
try to use this sync endpoint allows for seamless transitions between
altset 2 and altset 3. If any errors happen, the behavior remains the same.
$ more /proc/asound/card1/stream0
M-Audio Transit USB at usb-0000:00:14.0-2, full speed : USB Audio
Playback:
Status: Stop
Interface 1
Altset 1
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (ADAPTIVE)
Rates: 48001 - 96000 (continuous)
Interface 1
Altset 2
Format: S24_3LE
Channels: 2
Endpoint: 3 OUT (NONE)
Rates: 8000 - 48000 (continuous)
Interface 1
Altset 3
Format: S16_LE
Channels: 2
Endpoint: 3 OUT (ASYNC)
Rates: 8000 - 48000 (continuous)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a transition occurs between alternate settings that do not use the
same synchronization method, the substream pointers were not reset.
This prevents audio from being played during the second transition.
Identified and tested with M-Audio Transit device
(0763:2006 Midiman M-Audio Transit)
Details of the issue:
First playback to adaptive endpoint:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
[ 3169.297556] usb 1-2: setting usb interface 1:1
[ 3169.297568] usb 1-2: Creating new playback data endpoint #3
[ 3169.298563] usb 1-2: Setting params for ep #3 (type 0, 3 urbs), ret=0
[ 3169.298574] usb 1-2: Starting data EP @ffff880035fc8000
first playback to asynchronous endpoint:
$ aplay -Dhw:1,0 ~/16_48.wav
Playing WAVE '/home/plb/16_48.wav' : Signed 16 bit Little Endian,
Rate 48000 Hz, Stereo
[ 3204.520251] usb 1-2: setting usb interface 1:3
[ 3204.520264] usb 1-2: Creating new playback data endpoint #3
[ 3204.520272] usb 1-2: Creating new capture sync endpoint #83
[ 3204.521162] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3204.521177] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
[ 3204.521182] usb 1-2: Starting data EP @ffff880035fce000
[ 3204.521204] usb 1-2: Starting sync EP @ffff8800bd616000
second playback to adaptive endpoint: no audio and error on terminal:
$ aplay -Dhw:1,0 ~/24_96.wav
Playing WAVE '/home/plb/24_96.wav' : Signed 24 bit Little Endian in 3bytes,
Rate 96000 Hz, Stereo
aplay: pcm_write:1939: write error: Input/output error
[ 3239.483589] usb 1-2: setting usb interface 1:1
[ 3239.483601] usb 1-2: Re-using EP 3 in iface 1,1 @ffff880035fc8000
[ 3239.484590] usb 1-2: Setting params for ep #3 (type 0, 4 urbs), ret=0
[ 3239.484606] usb 1-2: Setting params for ep #83 (type 1, 4 urbs), ret=0
This last line shows that a sync endpoint is used when it shouldn't.
The sync endpoint is no longer valid and the pointers are corrupted
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
USB Audio Class version 2.0 supports three different parameter block sizes for
CUR requests, which are 1 byte (5.2.3.1 Layout 1 Parameter Block), 2 bytes
(5.2.3.2 Layout 2 Parameter Block) and 4 bytes (5.2.3.3 Layout 3 Parameter
Block). Use the correct size according to the specific control as it was
already done for UACv1. The allocated block size for control requests is
increased to support the 4 byte worst case.
Signed-off-by: Julian Scheel <julian@jusst.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the correct dB ranges of Bose Companion 5 and Drangonfly DAC 1.2.
Signed-off-by: Yao-Wen Mao <yaowen@google.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a monitor stream is active, the next PCM stream access results in
EBUSY error because of the check in line6_stream_start(). Fix this by
just skipping the submission of pending URBs when the stream is
already running instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=101431
Cc: <stable@vger.kernel.org> # v4.0+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Steinberg MI2 and MI4 interfaces are compatible with the USB class
audio spec, but the MIDI part of the devices is reported as a vendor
specific interface.
This patch adds entries to quirks-table.h to recognize the MIDI
endpoints. Audio functionality was already working and is unaffected by
this change.
Signed-off-by: Dominic Sacré <dominic.sacre@gmx.de>
Signed-off-by: Albert Huitsing <albert@huitsing.nl>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Changed ctl type for Input Gain Control and Input Gain Pad Control to
USB_MIXER_S16 as per section 5.2.5.7.11-12 in the USB Audio Class 2.0
definition.
Signed-off-by: Johan Rastén <johan@oljud.se>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for the XMOS based JLsounds I2SoverUSB board
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver worked around an error in the MAYA44 USB(+)'s mixer unit
descriptor by aborting before parsing the missing field. However,
aborting parsing too early prevented parsing of the other units
connected to this unit, so the capture mixer controls would be missing.
Fix this by moving the check for this descriptor error after the parsing
of the unit's input pins.
Reported-by: nightmixes <nightmixes@gmail.com>
Tested-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add mixer control names for the ESI Maya44 USB+ (which appears to be
identical width the AudioTrak Maya44 USB).
Reported-by: nightmixes <nightmixes@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This quirk allows us to avoid the noisy:
current rate 0 is different from the runtime rate
message every time playback starts. While USB DAC in the RR2150
supports reading the sample rate, it never returns a sample rate
other than zero in my observation with common sample rates.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix this from the logs:
usb 7-1: New USB device found, idVendor=046d, idProduct=08ca
...
usb 7-1: Warning! Unlikely big volume range (=3072), cval->res is probably wrong.
usb 7-1: [5] FU [Mic Capture Volume] ch = 1, val = 4608/7680/1
Signed-off-by: Wolfram Sang <wsa@the-dreams.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Spotted by sparse:
sound/usb/bcd2000/bcd2000.c:73:1: warning: symbol 'devices_used' was not declared. Should it be static?
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Microsoft LifeCam Studio (045e:0772) needs a similar quirk for
suppressing the wrong sample rate inquiry.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Roland SC-D70 reports its device class as vendor specific class and
the quirk QUIRK_AUDIO_FIXED_ENDPOINT was used for audio output.
In the quirks table the sampling rate was hard-coded to 44100 Hz
and therefore not worked when the sound module was in 48000 Hz mode.
In this change the quirk is changed to QUIRK_AUDIO_STANDARD_INTERFACE
but as the sound module reports incorrect bSubframeSize in its
descriptors, additional change is made in format.c to detect it and
to override it (which uses the existing code for Edirol SD-90).
Tested both when the sound module was in 44100 Hz mode and 48000 Hz
mode and both audio input and output. MIDI related part of the driver
is not touched.
Signed-off-by: Takamichi Horikawa <takamichiho@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds Microsoft LifeCam Cinema USB ID to the snd_usb_get_sample_rate_quirk list as the Lifecam Cinema does not appear to support getting the sample rate.
Fixes the issue where the LifeCam Cinema would wait for USB timeout and log the message "cannot get freq at ep 0x82" when accessed.
Addresses bug report https://bugzilla.kernel.org/show_bug.cgi?id=95961.
Signed-off-by: Adam Honse <calcprogrammer1@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3237"
Signed-off-by: Dmitry M. Fedin <dmitry.fedin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adding this quirk allows us to avoid the noisy
"cannot get freq at ep 0x1" message in dmesg output every time
playback starts.
This ought to affect other Benchmark DAC1 variations using the same
"Microchip Technology, Inc." chip as well, but I have only tested
with the "Pre" variant.
Signed-off-by: Eric Wong <normalperson@yhbt.net>
Cc: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device complies to the UAC1 standard but hides that fact with
proprietary descriptors. The autodetect quirk for Roland devices
catches the audio interface but misses the MIDI part, so a specific
quirk is needed.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Reported-by: Rafa Lafuente <rafalafuente@gmail.com>
Tested-by: Raphaël Doursenaud <raphael@doursenaud.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The usages of clamp() macro in sound/usb/line6/playback.c are just
wrong, the low and high values are swapped.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix problem where playback of Denon DA-300USB DAC sometimes does not
start and leads to error messages like "clock source 41 is not valid,
cannot use".
Solution: Treat this device the same as other Denon/Marantz devices in
sound/usb/quirks.c.
Tested with both PCM and DSD formats.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=93261
Signed-off-by: Frank C Guenther <bugzilla.frnkcg@spamgourmet.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adds a quirk to disable the check that the sample rate has been set correctly, as the Lifecam does not support getting the sample rate.
This means that we don't need to wait for the USB timeout when attempting to get the sample rate. Waiting for the timeout causes problems in some applications, which give up on the device acquisition process before it has had time to complete, resulting in no sound.
[minor tidy up by tiwai]
Signed-off-by: Joe Turner <joe@oampo.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The address cannot be negative so make it unsigned. Also, an unsigned
int is always sufficient for the length, so no need to overdo it with a
size_t. Finally, add in range checks to see if the values passed in
actually fit where they are used.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The firmware version is a single byte so have the variable type agree.
Since the address to this member is passed to the read function, using
an int is not even portable.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Put an upper bound on how long we will wait for the device to respond to
a read/write request (i.e., 100 milliseconds) and return an error if
this is reached.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The device indicates the result of a read/write operation by making the
status available on a subsequent request from the driver. This is not
ready immediately, though, so the driver is currently slamming the
device with hundreds of pointless requests before getting the expected
response. Add a two millisecond delay before each attempt. This is
approximately the behavior observed with version 4.2.7.1 of the Windows
driver.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The first URBs are submitted during the prepare stage. When .trigger is
called, the ALSA core saves a trigger tstamp that doesn't correspond to
the actual time when the samples are submitted. The trigger_tstamp is
now updated when the first data are submitted to avoid any time offsets.
A usb-specific trigger_tstamp_pending_update flag is used for now,
at some point the flag would need to move to the ALSA core, USB
is not the only interface where silent block transfers are programmed
as part of the prepare stage, with actual data enabled when .trigger
is called.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Provide a unique name for each driver instead of using "line6usb" for
all of them. This will allow for different configurations based on the
driver type.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
