Commit Graph

1094 Commits

Author SHA1 Message Date
Anssi Hannula 12a6116e66 ALSA: usb-audio: Add sample rate inquiry quirk for AudioQuest DragonFly
Avoid getting sample rate on AudioQuest DragonFly as it is unsupported
and causes noisy "cannot get freq at ep 0x1" messages when playback
starts.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-12-14 10:13:17 +01:00
Anssi Hannula 42e3121d90 ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly
AudioQuest DragonFly DAC reports a volume control range of 0..50
(0x0000..0x0032) which in USB Audio means a range of 0 .. 0.2dB, which
is obviously incorrect and would cause software using the dB information
in e.g. volume sliders to have a massive volume difference in 100..102%
range.

Commit 2d1cb7f658 ("ALSA: usb-audio: add dB range mapping for some
devices") added a dB range mapping for it with range 0..50 dB.

However, the actual volume mapping seems to be neither linear volume nor
linear dB scale, but instead quite close to the cubic mapping e.g.
alsamixer uses, with a range of approx. -53...0 dB.

Replace the previous quirk with a custom dB mapping based on some basic
output measurements, using a 10-item range TLV (which will still fit in
alsa-lib MAX_TLV_RANGE_SIZE).

Tested on AudioQuest DragonFly HW v1.2. The quirk is only applied if the
range is 0..50, so if this gets fixed/changed in later HW revisions it
will no longer be applied.

v2: incorporated Takashi Iwai's suggestion for the quirk application
method

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-12-14 10:13:17 +01:00
Clemens Ladisch a91e627e3f ALSA: usb-audio: work around CH345 input SysEx corruption
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>
2015-11-16 08:59:29 +01:00
Clemens Ladisch 1ca8b20130 ALSA: usb-audio: prevent CH345 multiport output SysEx corruption
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>
2015-11-16 08:59:24 +01:00
Clemens Ladisch 98d362becb ALSA: usb-audio: add packet size quirk for the Medeli DD305
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-11-16 08:59:09 +01:00
Jurgen Kramer 16771c7c70 ALSA: usb: Add native DSD support for Aune X1S
This patch adds native DSD support for the Aune X1S 32BIT/384 DSD DAC

Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-11-09 14:14:47 +01:00
Ricard Wanderlof 9fa5cf8c54 ALSA: USB-audio: Remove mixer entry from Zoom R16/24 quirk
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>
2015-10-19 12:38:10 +02:00
Ricard Wanderlof 759c90fe01 ALSA: USB-audio: Adjust max packet size calculation for tx_length_quirk
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>
2015-10-19 12:38:10 +02:00
Ricard Wanderlof e057044677 ALSA: USB-audio: Add quirk for Zoom R16/24 playback
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>
2015-10-19 12:38:09 +02:00
Ricard Wanderlof b97a936910 ALSA: USB-audio: Add offset parameter to copy_to_urb()
Preparation for adding Zoom R16/24 quirk.
No functional change.

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19 12:38:08 +02:00
Ricard Wanderlof 5cf310e976 ALSA: USB-audio: Break out creation of silent urbs from prepare_outbound_urb()
Refactoring in preparation for adding Zoom R16/24 quirk.
No functional change.

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19 12:38:08 +02:00
Ricard Wanderlof 4c4e4391b8 ALSA: USB-audio: Also move out hwptr_done wrap from prepare_playback_urb()
Refactoring in preparation for adding Zoom R16/24 quirk.
No functional change.

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19 12:38:07 +02:00
Ricard Wanderlof 07a40c2fc6 ALSA: USB-audio: Break out copying to urb from prepare_playback_urb()
Refactoring in preparation for adding Zoom R16/24 quirk.
No functional change.

Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-10-19 12:38:06 +02:00
Ricard Wanderlof dab9981756 ALSA: USB-audio: Add support for Novation Nocturn MIDIcontrol surface
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>
2015-10-16 14:28:59 +02:00
Ricard Wanderlof ab30965d9b ALSA: usb-audio: Fix max packet size calculation for USB audio
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>
2015-10-13 11:40:44 +02:00
Keith A. Milner ac77423609 ALSA: usb-audio: Allow any MIDI endpoint to drive use of interrupt transfer on newer Roland devices
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>
2015-10-11 18:18:59 +02:00
Dan Carpenter e87359efca ALSA: usb-audio: harmless underflow in snd_audigy2nx_led_put()
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>
2015-09-28 14:33:03 +02:00
Johan Rastén 5ee20bc792 ALSA: usb-audio: Change internal PCM order
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>
2015-09-07 10:57:27 +02:00
Yao-Wen Mao 6aa6925cad ALSA: usb-audio: correct the value cache check.
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>
2015-08-28 10:38:25 +02:00
Takashi Iwai 0662292aec ALSA: usb-audio: Handle normal and auto-suspend equally
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>
2015-08-26 16:12:25 +02:00
Takashi Iwai a6da499b76 ALSA: usb-audio: Replace probing flag with active refcount
We can use active refcount for preventing autopm during probe.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-08-26 15:40:18 +02:00
Takashi Iwai 47ab154593 ALSA: usb-audio: Avoid nested autoresume calls
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>
2015-08-26 15:38:25 +02:00
Takashi Iwai 00833d70ca Merge branch 'for-linus' into for-next 2015-08-21 19:26:48 +02:00
Jurgen Kramer 9544f8b6e2 ALSA: usb: Add native DSD support for Gustard DAC-X20U
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>
2015-08-21 10:27:35 +02:00
Julian Scheel 9430e54789 ALSA: usb-audio: Recurse before saving terminal properties
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>
2015-08-19 18:05:13 +02:00
Takashi Iwai 9003ebb13f ALSA: usb-audio: Fix runtime PM unbalance
The fix for deadlock in PM in commit [1ee23fe07ee8: 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>
2015-08-19 14:57:51 +02:00
Pierre-Louis Bossart 395ae54bd8 ALSA: usb: handle descriptor with SYNC_NONE illegal value
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>
2015-08-16 08:48:47 +02:00
Pierre-Louis Bossart 630184477e ALSA: usb: fix corrupted pointers due to interface setting change
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>
2015-08-16 08:48:35 +02:00
Julian Scheel bc18e31c30 ALSA: usb-audio: Fix parameter block size for UAC2 control requests
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>
2015-08-14 16:26:50 +02:00
Yao-Wen Mao 2d1cb7f658 ALSA: usb-audio: add dB range mapping for some devices
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>
2015-07-29 09:28:02 +02:00
Takashi Iwai 4d0e677523 ALSA: line6: Fix -EBUSY error during active monitoring
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>
2015-07-14 15:19:37 +02:00
Dominic Sacré 0689a86ae8 ALSA: usb-audio: Add MIDI support for Steinberg MI2/MI4
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>
2015-07-01 17:29:40 +02:00
Johan Rastén 27c41dad3a ALSA: usb-audio: Set correct type for some UAC2 mixer controls.
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>
2015-06-11 11:57:35 +02:00
Takashi Iwai 8654844cf5 Merge branch 'for-linus' into for-next
Resolve the non-trivial conflict due to the hdac regmap API changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-06-09 07:22:26 +02:00
Jurgen Kramer 3b7e5c7e36 ALSA: usb-audio: add native DSD support for JLsounds I2SoverUSB
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>
2015-06-08 11:22:21 +02:00
Clemens Ladisch ea114fc27d ALSA: usb-audio: fix missing input volume controls in MAYA44 USB(+)
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>
2015-06-03 11:58:15 +02:00
Clemens Ladisch 044bddb9ca ALSA: usb-audio: add MAYA44 USB+ mixer control names
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>
2015-06-03 11:57:51 +02:00
Eric Wong 2f80b2958a ALSA: usb-audio: don't try to get Outlaw RR2150 sample rate
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>
2015-05-30 14:14:40 +02:00
Wolfram Sang 1ef9f05835 ALSA: usb-audio: Add mic volume fix quirk for Logitech Quickcam Fusion
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>
2015-05-29 12:57:49 +02:00
Takashi Iwai 984a854705 Merge branch 'for-linus' into for-next
Merge back the latest HD-audio stuff for further development.
2015-05-29 10:27:50 +02:00
Takashi Iwai 574d69c27b ALSA: bcd2000: Make local data static
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>
2015-05-26 13:00:01 +02:00
Vittorio G (VittGam) ae425bb2a0 ALSA: usb-audio: Add quirk for MS LifeCam HD-3000
Microsoft LifeCam HD-3000 (045e:0779) needs a similar quirk for
suppressing the unsupported sample rate inquiry.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=98481
Cc: <stable@vger.kernel.org>
Signed-off-by: Vittorio Gambaletta <linuxbugs@vittgam.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-05-24 08:26:55 +02:00
Takashi Iwai fa94b0d725 ALSA: usb-audio: Add quirk for MS LifeCam Studio
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>
2015-05-19 10:46:49 +02:00
Takamichi Horikawa 6d1f2f6056 ALSA: usb-audio: Fix audio output on Roland SC-D70 sound module
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>
2015-04-21 07:59:10 +02:00
Takashi Iwai 9a4f35865f Merge branch 'for-next' into for-linus 2015-04-13 10:23:18 +02:00
Adam Honse eef0342cf3 ALSA: usb-audio: Don't attempt to get Microsoft Lifecam Cinema sample rate
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>
2015-04-12 09:08:42 +02:00
Dmitry M. Fedin 3dc8523fa7 ALSA: usb - Creative USB X-Fi Pro SB1095 volume knob support
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>
2015-04-09 17:20:39 +02:00
Takashi Iwai 0a59983873 Merge branch 'for-linus' into for-next
Back merge HD-audio quirks to for-next branch, so that we can apply
a couple of more quirks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-04-08 11:30:49 +02:00
Eric Wong 9fc88ad6fd ALSA: usb-audio: don't try to get Benchmark DAC1 sample rate
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>
2015-04-04 14:07:56 +02:00
Takashi Iwai 34e72afe73 Merge branch 'for-linus' into for-next 2015-03-16 14:48:05 +01:00