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  1. 04 3月, 2016 1 次提交
  3. 30 1月, 2016 1 次提交
  5. 29 1月, 2016 2 次提交
  7. 26 1月, 2016 1 次提交
  9. 13 1月, 2016 1 次提交
  11. 12 1月, 2016 1 次提交
    • T
      ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect · 5c06d68b
      由 Takashi Iwai 提交于 
      ALSA PCM may still have a leftover instance after disconnection and
it delays its release.  The problem is that the PCM close code path of
USB-audio driver has a call of snd_usb_autosuspend().  This involves
with the call of usb_autopm_put_interface() and it may lead to a
kernel Oops due to the NULL object like:

 BUG: unable to handle kernel NULL pointer dereference at 0000000000000190
 IP: [<ffffffff815ae7ef>] usb_autopm_put_interface+0xf/0x30 PGD 0
 Call Trace:
  [<ffffffff8173bd94>] snd_usb_autosuspend+0x14/0x20
  [<ffffffff817461bc>] snd_usb_pcm_close.isra.14+0x5c/0x90
  [<ffffffff8174621f>] snd_usb_playback_close+0xf/0x20
  [<ffffffff816ef58a>] snd_pcm_release_substream.part.36+0x3a/0x90
  [<ffffffff816ef6b3>] snd_pcm_release+0xa3/0xb0
  [<ffffffff816debb0>] snd_disconnect_release+0xd0/0xe0
  [<ffffffff8114d417>] __fput+0x97/0x1d0
  [<ffffffff8114d589>] ____fput+0x9/0x10
  [<ffffffff8109e452>] task_work_run+0x72/0x90
  [<ffffffff81088510>] do_exit+0x280/0xa80
  [<ffffffff8108996a>] do_group_exit+0x3a/0xa0
  [<ffffffff8109261f>] get_signal+0x1df/0x540
  [<ffffffff81040903>] do_signal+0x23/0x620
  [<ffffffff8114c128>] ? do_readv_writev+0x128/0x200
  [<ffffffff810012e1>] prepare_exit_to_usermode+0x91/0xd0
  [<ffffffff810013ba>] syscall_return_slowpath+0x9a/0x120
  [<ffffffff817587cd>] ? __sys_recvmsg+0x5d/0x70
  [<ffffffff810d2765>] ? ktime_get_ts64+0x45/0xe0
  [<ffffffff8115dea0>] ? SyS_poll+0x60/0xf0
  [<ffffffff818d2327>] int_ret_from_sys_call+0x25/0x8f

We have already a check of disconnection in snd_usb_autoresume(), but
the check is missing its counterpart.  The fix is just to put the same
check in snd_usb_autosuspend(), too.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=109431
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      5c06d68b
  13. 11 1月, 2016 1 次提交
  15. 22 12月, 2015 1 次提交
  17. 14 12月, 2015 2 次提交
    • A
      ALSA: usb-audio: Add sample rate inquiry quirk for AudioQuest DragonFly · 12a6116e
      由 Anssi Hannula 提交于 
      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: NAnssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      12a6116e
    • A
      ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly · 42e3121d
      由 Anssi Hannula 提交于 
      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 2d1cb7f6 ("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: NAnssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      42e3121d
  19. 11 12月, 2015 1 次提交
  21. 08 12月, 2015 1 次提交
  23. 03 12月, 2015 1 次提交
  25. 22 11月, 2015 1 次提交
  27. 16 11月, 2015 4 次提交
    • C
      ALSA: ua101: replace le16_to_cpu() with usb_endpoint_maxp() · 3c7a0935
      由 Cheah Kok Cheong 提交于 
      Commit 939f325f ("usb: add usb_endpoint_maxp() macro") and commit
29cc8897 ("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: NCheah Kok Cheong <thrust73@gmail.com>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      3c7a0935
    • C
      ALSA: usb-audio: work around CH345 input SysEx corruption · a91e627e
      由 Clemens Ladisch 提交于 
      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: NClemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      a91e627e
    • C
      ALSA: usb-audio: prevent CH345 multiport output SysEx corruption · 1ca8b201
      由 Clemens Ladisch 提交于 
      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: NClemens Ladisch <clemens@ladisch.de>
Cc: stable@vger.kernel.org
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      1ca8b201
    • C
      ALSA: usb-audio: add packet size quirk for the Medeli DD305 · 98d362be
      由 Clemens Ladisch 提交于 
      Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      98d362be
  29. 09 11月, 2015 1 次提交
  31. 19 10月, 2015 7 次提交
  33. 16 10月, 2015 1 次提交
  35. 13 10月, 2015 1 次提交
    • R
      ALSA: usb-audio: Fix max packet size calculation for USB audio · ab30965d
      由 Ricard Wanderlof 提交于 
      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: NRicard Wanderlof <ricardw@axis.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      ab30965d
  37. 12 10月, 2015 1 次提交
  39. 28 9月, 2015 1 次提交
  41. 07 9月, 2015 1 次提交
    • J
      ALSA: usb-audio: Change internal PCM order · 5ee20bc7
      由 Johan Rastén 提交于 
      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: NJohan Rastén <johan@oljud.se>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      5ee20bc7
  43. 28 8月, 2015 1 次提交
  45. 26 8月, 2015 3 次提交
    • T
      ALSA: usb-audio: Handle normal and auto-suspend equally · 0662292a
      由 Takashi Iwai 提交于 
      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: NTakashi Iwai <tiwai@suse.de>
      0662292a
    • T
      ALSA: usb-audio: Replace probing flag with active refcount · a6da499b
      由 Takashi Iwai 提交于 
      We can use active refcount for preventing autopm during probe.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      a6da499b
    • T
      ALSA: usb-audio: Avoid nested autoresume calls · 47ab1545
      由 Takashi Iwai 提交于 
      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: 9003ebb1 ('ALSA: usb-audio: Fix runtime PM unbalance')
Reported-and-tested-by: NAlexnader Kuleshov <kuleshovmail@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      47ab1545
  47. 21 8月, 2015 1 次提交
  49. 20 8月, 2015 1 次提交
  51. 19 8月, 2015 1 次提交
    • T
      ALSA: usb-audio: Fix runtime PM unbalance · 9003ebb1
      由 Takashi Iwai 提交于 
      The fix for deadlock in PM in commit [1ee23fe0: 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: NHans Yang <hansy@nvidia.com>
Fixes: 1ee23fe0 ('ALSA: usb-audio: Fix deadlocks at resuming')
Cc: <stable@vger.kernel.org> [v3.15+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      9003ebb1
  53. 16 8月, 2015 1 次提交
    • P
      ALSA: usb: handle descriptor with SYNC_NONE illegal value · 395ae54b
      由 Pierre-Louis Bossart 提交于 
      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: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      395ae54b