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  1. 26 8月, 2015 1 次提交
    • 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
  3. 21 8月, 2015 1 次提交
  5. 20 8月, 2015 1 次提交
  7. 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
  9. 16 8月, 2015 2 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
    • P
      ALSA: usb: fix corrupted pointers due to interface setting change · 63018447
      由 Pierre-Louis Bossart 提交于 
      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: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      63018447
  11. 14 8月, 2015 1 次提交
  13. 29 7月, 2015 1 次提交
  15. 14 7月, 2015 1 次提交
  17. 01 7月, 2015 1 次提交
  19. 11 6月, 2015 1 次提交
  21. 08 6月, 2015 1 次提交
  23. 03 6月, 2015 2 次提交
  25. 30 5月, 2015 1 次提交
  27. 29 5月, 2015 1 次提交
  29. 26 5月, 2015 1 次提交
  31. 24 5月, 2015 1 次提交
  33. 19 5月, 2015 1 次提交
  35. 21 4月, 2015 1 次提交
    • T
      ALSA: usb-audio: Fix audio output on Roland SC-D70 sound module · 6d1f2f60
      由 Takamichi Horikawa 提交于 
      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: NTakamichi Horikawa <takamichiho@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      6d1f2f60
  37. 12 4月, 2015 1 次提交
  39. 09 4月, 2015 1 次提交
  41. 04 4月, 2015 1 次提交
  43. 12 3月, 2015 1 次提交
  45. 05 3月, 2015 1 次提交
  47. 04 3月, 2015 1 次提交
  49. 18 2月, 2015 1 次提交
  51. 17 2月, 2015 1 次提交
    • J
      ALSA: usb-audio: Don't attempt to get Lifecam HD-5000 sample rate · b62b9980
      由 Joe Turner 提交于 
      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: NJoe Turner <joe@oampo.co.uk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      b62b9980
  53. 12 2月, 2015 1 次提交
  55. 11 2月, 2015 5 次提交
  57. 09 2月, 2015 1 次提交
  59. 08 2月, 2015 3 次提交
  61. 06 2月, 2015 2 次提交