The snd_rawmidi_global_ops structures are never modified, so declare them
as const.
Done with the help of Coccinelle.
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 939f325f4a ("usb: add usb_endpoint_maxp() macro") and commit
29cc88979a ("USB: use usb_endpoint_maxp() instead of le16_to_cpu()")
introduced a new helper macro. This trivial patch convert remaining
users found in ua101 driver.
Signed-off-by: Cheah Kok Cheong <thrust73@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One of the many faults of the QinHeng CH345 USB MIDI interface chip is
that it does not handle received SysEx messages correctly -- every second
event packet has a wrong code index number, which is the one from the last
seen message, instead of 4. For example, the two messages "FE F0 01 02 03
04 05 06 07 08 09 0A 0B 0C 0D 0E F7" result in the following event
packets:
correct: CH345:
0F FE 00 00 0F FE 00 00
04 F0 01 02 04 F0 01 02
04 03 04 05 0F 03 04 05
04 06 07 08 04 06 07 08
04 09 0A 0B 0F 09 0A 0B
04 0C 0D 0E 04 0C 0D 0E
05 F7 00 00 05 F7 00 00
A class-compliant driver must interpret an event packet with CIN 15 as
having a single data byte, so the other two bytes would be ignored. The
message received by the host would then be missing two bytes out of six;
in this example, "F0 01 02 03 06 07 08 09 0C 0D 0E F7".
These corrupted SysEx event packages contain only data bytes, while the
CH345 uses event packets with a correct CIN value only for messages with
a status byte, so it is possible to distinguish between these two cases by
checking for the presence of this status byte.
(Other bugs in the CH345's input handling, such as the corruption resulting
from running status, cannot be worked around.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The CH345 USB MIDI chip has two output ports. However, they are
multiplexed through one pin, and the number of ports cannot be reduced
even for hardware that implements only one connector, so for those
devices, data sent to either port ends up on the same hardware output.
This becomes a problem when both ports are used at the same time, as
longer MIDI commands (such as SysEx messages) are likely to be
interrupted by messages from the other port, and thus to get lost.
It would not be possible for the driver to detect how many ports the
device actually has, except that in practice, _all_ devices built with
the CH345 have only one port. So we can just ignore the device's
descriptors, and hardcode one output port.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
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>
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>
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>
It is unlikely this function would ever be used in a context without a
pointer to a `struct usb_line6_toneport', so grab the device type from
it rather than having the caller do it.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a predicate for testing if the device supports source selection to
make the conditional logic around this a bit cleaner.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
line6_start_timer passes an unsigned int as argument to be used in mod_timer
which is then used by mod_timer as unsigned long, this just fixes up the
argument type. This change helps make static code checkers happy.
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is only an API consolidation and should make things more readable
it replaces var * HZ / 1000 by msecs_to_jiffies(var).
Signed-off-by: Nicholas Mc Guire <hofrat@osadl.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use of this function ended with commits 3e58c868db ("staging: line6:
drop midi_mask_receive") and af89d2897a ("staging: line6: drop
midi_mask_transmit".)
[Removed the corresponding line in midibuf.h, too -- tiwai]
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This function has not been used since merging the driver into the kernel
(and a good while before that.)
[Removed the corresponding line in midibuf.h, too -- tiwai]
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most of them are rather relevant with the definitions in driver.h,
and there are only a few lines, so just rip it off.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just reformatting the comments and typos fixed, no functional
changes. Particularly,
- avoid the kerneldoc marker "/**",
- reduce multiple comment lines into single lines,
- corrected wrongly referred function names
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The user-space API definition for usb_stream stuff should be moved
to include/uapi/sound to be exposed publicly.
While we're at it, add the missing ifdef guard for double inclusion,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Both playback and capture callbacks are identical, so let's merge
them.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The current code deals with the stream start / stop solely via
line6_pcm_acquire() and line6_pcm_release(). This was (supposedly)
intended to avoid the races, but it doesn't work as expected. The
concurrent acquire and release calls can be performed without proper
protections, thus this might result in memory corruption.
Furthermore, we can't take a mutex to protect the whole function
because it can be called from the PCM trigger callback that is an
atomic context. Also spinlock isn't appropriate because the function
allocates with kmalloc with GFP_KERNEL. That is, these function just
lead to singular problems.
This is an attempt to reduce the existing races. First off, separate
both the stream buffer management and the stream URB management. The
former is protected via a newly introduced state_mutex while the
latter is protected via each line6_pcm_stream lock.
Secondly, the stream state are now managed in opened and running bit
flags of each line6_pcm_stream. Not only this a bit clearer than
previous combined bit flags, this also gives a better abstraction.
These rewrites allows us to make common hw_params and hw_free
callbacks for both playback and capture directions.
For the monitor and impulse operations, still line6_pcm_acquire() and
line6_pcm_release() are used. They call internally the corresponding
functions for both playback and capture streams with proper lock or
mutex. Unlike the previous versions, these function don't take the
bit masks but the only single type value. Also they are supposed to
be applied only as duplex operations.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clearing prev_fsize in line6_pcm_acquire() is pretty racy.
This can be called at any time while the stream is being played.
Rather better to clear prev_fbuf and prev_fsize at the proper place
like the stream stop for capture, and just after copying the monitor /
impulse data inside the spinlock.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The impulse and monitor handling in submit_audio_out_urb() isn't
protected thus this can be racy with the capture stream handling.
This patch extends the range to protect via each stream's spinlock
(now the whole submit_audio_*_urb() are covered), and take the capture
stream lock additionally for the impulse and monitor handling part.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the check of multi configurations before snd_card_new() as a
short path, and reduce superfluous pointer references.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of allocating the private data individually in each driver's
probe at first, let snd_card_new() allocate the data that is called in
line6_probe(). This simplifies the primary probe functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interface argument is used just for retrieving the assigned
device, which can be already found in line6->ifcdev. Drop them from
the callbacks. Also, pass the usb id to private_init so that the
driver can deal with it there. This is a preliminary work for the
further cleanup to move the whole allocation into driver.c.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A minor optimization; while pausing, the driver just copies the zero
that doesn't need any volume changes.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM stream buffer allocation and free are identical for both
playback and capture streams. Provide single helper functions.
These are used only in pcm.c, thus they can be even static.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The codes to unlink and sync URBs are identical for both playback and
capture streams. Consolidate to single helper functions.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce a new line6_pcm_stream structure and group individual
fields of snd_line6_pcm struct to playback and capture groups.
This patch itself just does rename and nothing else. More
meaningful cleanups based on these fields shuffling will follow.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the problem still really remains, we should fix it instead of
papering over it like this...
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using a decremental loop without particular reasons worsens the
readability a lot. Use incremental loops instead.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The trigger callback is already spinlocked, so we need no more lock
here (even for the linked substreams). Let's drop it.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
line6_pcm_acquire() tries to restore the newly obtained resources at
the error path. But some flags aren't recorded and released properly
when the corresponding buffer is already present. These bits have to
be cleared in the error recovery, too.
Also, "flags_final" can be initialized to zero since we pass only the
subset of "channels" bits.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixed a few places using bits OR wrongly for condition checks.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The midi_transmit_lock is used always inside the send_urb_lock, thus
it doesn't play any role. Let's kill it. Also, rename
"send_urb_lock" as a more simple name "lock" since this is the only
lock for midi.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The function isn't used any longer after rewriting from sysfs to leds
class in toneport.c.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix memory leak at probe error path by rearranging the call order in
line6_destruct() so that the common destructor is always called.
Also this simplifies the error path to a single goto label.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of non-standard sysfs, reimplement the LED controls on
TonePort as LED class devices.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Currently disconnect callback is used as a driver's destructor, and
this has to be called not only at the disconnection time but also at
the error paths during probe.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
... so that timer_del_sync() in the destructor can be called safely at
any time. Also move the mod_timer() call in toneport_setup(), which
is a bit clearer place.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The interface and driver objects are always set when callbacks are
called. Drop such superfluous NULL checks in init and disconnect
calls of each driver.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's utterly unsafe to proceed further the disconnect procedure if the
assigned usbdev is inconsistent with the expected object. Better to
put a WARN_ON() for more cautions and abort immediately.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver version string was removed in an ealier commit for being
useless. These are equally useless.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The correct spelling includes the space. Fix this in strings and
comments that refer to the manufacturer.
Signed-off-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Call line6_pcm_disconnect() at disconnect to make sure that all URBs
are cleared. Also reduce the superfluous snd_pcm_stop() calls from
the function (and remove the unused function) since the streams are
guaranteed to be stopped at this point via snd_card_disconnect().
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Calling line6_pcm_disconnect() at suspend callback is superfluous and
rather confusing. Let's get rid of it.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Such a debug is needed in the core code, not in each lowlevel driver.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is rather useless for a driver that has been already merged into
the official tree.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is a fairly big rewrite regarding the card resource management in
line6 drivers:
- The card creation is moved into line6_probe(). This adds the global
destructor to private_free, so that each driver doesn't have to call
it any longer.
- The USB disconnect callback handles the card release, thus each
driver needs to concentrate on only its own resources. No need to
snd_card_*() call in the destructor.
- Fix the potential stall in disconnection by removing
snd_card_free(). It's replaced with snd_card_free_when_closed()
for asynchronous release.
- The only remaining operation for the card in each driver is the call
of snd_card_register(). All the rest are dealt in the common module
by itself.
- These ended up with removal of audio.[ch] as a result of a reduction
of one layer. Each driver just needs to call line6_probe().
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The PCM trigger callback is guaranteed to be called already in
spinlock / irq-disabled context.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The line6 drivers don't support the full resume although they set
SNDRV_PCM_INFO_RESUME. These flags have to be dropped to inform
properly to the user-space.
Also, drop the CONFIG_PM in trigger callbacks, too, which are rather
superfluous.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Like the previous fix for PCM, attach the card-specific resource into
rawmidi->private_data instead of handling in a snd_device object.
This simplifies the code and structure.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of handling the card-specific resource in snd_device, attach
it into pcm->private_data and release it directly in private_free.
This simplifies the code and structure.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of sysfs and the conditional build with Kconfig, implement the
handling of the impulse response controls via control API, and always
enable the build. Two new controls, "Impulse Response Volume" and
"Impulse Response Period" are added as a replacement for the former
sysfs files.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Split to each individual driver for POD, PODHD, TonePort and Variax
with a core LINE6 helper module. The new modules follow the standard
ALSA naming rule with snd prefix: snd-usb-pod, snd-usb-podhd,
snd-usb-toneport and snd-usb-variax, together with the corresponding
CONFIG_SND_USB_* Kconfig items.
Tested-by: Chris Rorvick <chris@rorvick.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Promote line6 driver from staging to sound/usb/line6 directory, and
maintain through sound subsystem tree.
This commit just moves the code and adapts Makefile / Kconfig.
The further renames and misc cleanups will follow.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Akai MPC Element incorrectly reports its bInterfaceClass as 255, but
otherwise implements the USB MIDI spec correctly.
This adds a quirks-table.h entry which allows the device to be
recognized as a standard USB MIDI device.
Signed-off-by: Paul Bonser <misterpib@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit 897c329bc ("ALSA: usb: caiaq: check for cdev->n_streams > 1")
introduced a safety check to protect against bogus data provided by
devices. However, the n_streams variable is already divided by
CHANNELS_PER_STREAM, so the correct check is 'n_streams > 0'.
Fix this to un-break support for stereo devices.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Cc: stable@kernel.org [v3.18+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Here are a few fixes that have landed after the previous pull
request. All are driver specific fixes including:
- error/int value fixes in OXFW,
- Intel Skylake HD-audio HDMI codec support,
- Additional HD-audio Realtek codecs and AD1986A codec fixes/quirks,
- a few more DSD support and a quirk for Arcam rPAC in usb-audio,
- a typo fix for Scarlett 6i6,
- fixes for new ASIHPI firmware,
- ASoC Exynos7 cleanups,
- Intel ACPI support, and
- a fix for PCM512 register cache sync.
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Merge tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few fixes that have landed after the previous pull request.
All are driver specific fixes including:
- error/int value fixes in OXFW,
- Intel Skylake HD-audio HDMI codec support,
- Additional HD-audio Realtek codecs and AD1986A codec fixes/quirks,
- a few more DSD support and a quirk for Arcam rPAC in usb-audio,
- a typo fix for Scarlett 6i6,
- fixes for new ASIHPI firmware,
- ASoC Exynos7 cleanups,
- Intel ACPI support, and
- a fix for PCM512 register cache sync"
* tag 'sound-fix-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (24 commits)
ALSA: usb-audio: extend KEF X300A FU 10 tweak to Arcam rPAC
ALSA: hda/realtek - New codec support for ALC298
ALSA: asihpi: update to HPI version 4.14
ALSA: asihpi: increase tuner pad cache size
ALSA: asihpi: relax firmware version check
ALSA: usb-audio: Fix Scarlett 6i6 initialization typo
ALSA: hda - Add quirk for Packard Bell EasyNote MX65
ALSA: usb-audio: add native DSD support for Matrix Audio DACs
ALSA: hda/realtek - New codec support for ALC256
ALSA: hda/realtek - Add new Dell desktop for ALC3234 headset mode
ASoC: Intel: fix possible acpi enumeration panic
ALSA: hda/hdmi - apply Haswell fix-ups to Skylake display codec
ASoC: Intel: fix return value check in sst_acpi_probe()
ALSA: hda - Make add_stereo_mix_input flag tristate
ALSA: hda - Create capture source ctls when stereo mix input is added
ALSA: hda - Fix typos in snd_hda_get_int_hint() kerneldoc comments
ALSA: hda - add codec ID for Skylake display audio codec
ALSA: oxfw: some signedness bugs
ALSA: oxfw: fix detect_loud_models() return value
ASoC: rt5677: add REGMAP_I2C and REGMAP_IRQ dependency
...
The Arcam rPAC seems to have the same problem - whenever anything
(alsamixer, udevd, 3.9+ kernel from 60af3d037e, ..) attempts to
access mixer / control interface of the card, the firmware "locks up"
the entire device, resulting in
SNDRV_PCM_IOCTL_HW_PARAMS failed (-5): Input/output error
from alsa-lib.
Other operating systems can somehow read the mixer (there seems to be
playback volume/mute), but any manipulation is ignored by the device
(which has hardware volume controls).
Cc: <stable@vger.kernel.org>
Signed-off-by: Jiri Jaburek <jjaburek@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The num_controls field was incorrectly set to 0 causing 6i6 to not be
initialized. Set this to 9.
Reported-and-tested-by: Mark Roberts <sunifiram@gmail.com>
Signed-off-by: Chris J Arges <chris.j.arges@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds native DSD support for two XMOS based DACs from Matrix Audio:
- X-Sabre
- Mini-i Pro
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
* ALSA core
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
* USB-audio
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
* FireWire
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
* HD-audio
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
* ASoC
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
* Others
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle
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Merge tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became a fairly large pull request. In addition to the usual
driver updates / fixes, there have been a high amount of cleanups in
ASoC area, as well as control API helpers and kernel documentations
fixes touching through the whole tree.
In the driver side, the biggest changes are the support for new Intel
SoC found on new x86 machines, and the updates of FireWire dice and
oxfw drivers.
Some remarkable items are below:
ALSA core:
- PCM mmap code cleanup, removal of arch-dependent codes
- PCM xrun injection support
- PCM hwptr tracepoint support
- Refactoring of snd_pcm_action(), simplification of PCM locking
- Robustified sequecner auto-load functionality
- New control API helpers and lots of cleanups along with them
- Lots of kerneldoc fixes and cleanups
USB-audio:
- The mixer resume code was largely rewritten, and the devices with
quirks are resumed properly.
- New hardware support: Focusrite Scarlett, Digidesign Mbox1,
Denon/Marantz DACs, Zoom R16/24
FireWire:
- DICE driver updates with better duplex and sync support, including
MIDI support
- New OXFW driver for Oxford Semiconductor FW970/971 chipset,
including the previous LaCie Speakers device. Fullduplex and MIDI
support included as well as DICE driver.
HD-audio:
- Refactoring the driver-caps quirk handling in snd-hda-intel
- More consistent control names representing the topology better
- Fixups: HP mute LED with ALC268 codec, Ideapad S210 built-in mic
fix, ASUS Z99He laptop EAPD
ASoC:
- Conversion of AC'97 drivers to use regmap, bringing us closer to
the removal of the ASoC level I/O code
- Clean up a lot of old drivers that were open coding things that
have subsequently been implemented in the core
- Some DAPM performance improvements
- Removal of the now seldom used CODEC mutex
- Lots of updates for the newer Intel SoC support, including support
for the DSP and some Cherrytrail and Braswell machine drivers
- Support for Samsung boards using rt5631 as the CODEC
- Removal of the obsolete AFEB9260 machine driver
- Driver support for the TI TS3A227E headset driver used in some
Chrombeooks
Others:
- ASIHPI driver update and cleanups
- Lots of dev_*() printk conversions
- Lots of trivial cleanups for the codes spotted by Coccinelle"
* tag 'sound-3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (594 commits)
ALSA: pcxhr: NULL dereference on probe failure
ALSA: lola: NULL dereference on probe failure
ALSA: hda - Add "eapd" model string for AD1986A codec
ALSA: hda - Add EAPD fixup for ASUS Z99He laptop
ALSA: oxfw: Add hwdep interface
ALSA: oxfw: Add support for capture/playback MIDI messages
ALSA: oxfw: add support for capturing PCM samples
ALSA: oxfw: Add support AMDTP in-stream
ALSA: oxfw: Add support for Behringer/Mackie devices
ALSA: oxfw: Change the way to start stream
ALSA: oxfw: Add proc interface for debugging purpose
ALSA: oxfw: Change the way to make PCM rules/constraints
ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation
ALSA: oxfw: Change the way to name card
ALSA: dice: Add support for MIDI capture/playback
ALSA: dice: Add support for capturing PCM samples
ALSA: dice: Support for non SYT-Match sampling clock source mode
ALSA: dice: Add support for duplex streams with synchronization
ALSA: dice: Change the way to start stream
ALSA: jack: Add dummy snd_jack_set_key() definition
...
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Merge tag 'media/v3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media
Pull media updates from Mauro Carvalho Chehab:
- Two new dvb frontend drivers: mn88472 and mn88473
- A new driver for some PCIe DVBSky cards
- A new remote controller driver: meson-ir
- One LIRC staging driver got rewritten and promoted to mainstream:
igorplugusb
- A new tuner driver (m88rs6000t)
- The old omap2 media driver got removed from staging. This driver
uses an old DMA API and it is likely broken on recent kernels.
Nobody cared enough to fix it
- Media bus format moved to a separate header, as DRM will also use the
definitions there
- mem2mem_testdev were renamed to vim2m, in order to use the same
naming convention taken by the other virtual test driver (vivid)
- Added a new driver for coda SoC (coda-jpeg)
- The cx88 driver got converted to use videobuf2 core
- Make DMABUF export buffer to work with DMA Scatter/Gather and Vmalloc
cores
- Lots of other fixes, improvements and cleanups on the drivers.
* tag 'media/v3.19-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media: (384 commits)
[media] mn88473: One function call less in mn88473_init() after error
[media] mn88473: Remove uneeded check before release_firmware()
[media] lirc_zilog: Deletion of unnecessary checks before vfree()
[media] MAINTAINERS: Add myself as img-ir maintainer
[media] img-ir: Don't set driver's module owner
[media] img-ir: Depend on METAG or MIPS or COMPILE_TEST
[media] img-ir/hw: Drop [un]register_decoder declarations
[media] img-ir/hw: Fix potential deadlock stopping timer
[media] img-ir/hw: Always read data to clear buffer
[media] redrat3: ensure dma is setup properly
[media] ddbridge: remove unneeded check before dvb_unregister_device()
[media] si2157: One function call less in si2157_init() after error
[media] tuners: remove uneeded checks before release_firmware()
[media] arm: omap2: rx51-peripherals: fix build warning
[media] stv090x: add an extra protetion against buffer overflow
[media] stv090x: Remove an unreachable code
[media] stv090x: Some whitespace cleanups
[media] em28xx: checkpatch cleanup: whitespaces/new lines cleanups
[media] si2168: add support for firmware files in new format
[media] si2168: debug printout for firmware version
...
