Commit Graph

1310 Commits

Author SHA1 Message Date
Takashi Iwai ebe2aa911f Merge branch 'topic/for-4.12' into for-next 2017-03-20 12:24:26 +01:00
Dave Jiang 11bac80004 mm, fs: reduce fault, page_mkwrite, and pfn_mkwrite to take only vmf
->fault(), ->page_mkwrite(), and ->pfn_mkwrite() calls do not need to
take a vma and vmf parameter when the vma already resides in vmf.

Remove the vma parameter to simplify things.

[arnd@arndb.de: fix ARM build]
  Link: http://lkml.kernel.org/r/20170125223558.1451224-1-arnd@arndb.de
Link: http://lkml.kernel.org/r/148521301778.19116.10840599906674778980.stgit@djiang5-desk3.ch.intel.com
Signed-off-by: Dave Jiang <dave.jiang@intel.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Ross Zwisler <ross.zwisler@linux.intel.com>
Cc: Theodore Ts'o <tytso@mit.edu>
Cc: Darrick J. Wong <darrick.wong@oracle.com>
Cc: Matthew Wilcox <mawilcox@microsoft.com>
Cc: Dave Hansen <dave.hansen@intel.com>
Cc: Christoph Hellwig <hch@lst.de>
Cc: Jan Kara <jack@suse.com>
Cc: Dan Williams <dan.j.williams@intel.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2017-02-24 17:46:54 -08:00
Takashi Iwai 7086b7b3d1 ALSA: usb-audio: Tidy up mixer_us16x08.c
A few more cleanups and improvements that have been overlooked:

- Use ARRAY_SIZE() macro appropriately
- Code shuffling for minor optimization
- Omit superfluous variable initializations
- Get rid of superfluous NULL checks
- Add const to snd_us16x08_control_params definitions

No functional changes.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-22 15:07:20 +01:00
Takashi Iwai e2810d76c5 ALSA: usb-audio: Fix memory leak and corruption in mixer_us16x08.c
There are a few places leaking memory and doing double-free in
mixer_us16x08.c.

The driver allocates a usb_mixer_elem_info object at each
add_new_ctl() call.  This has to be freed via kctl->private_free, but
currently this is done properly only for some controls.

Also, the driver allocates three external objects (comp_store,
eq_store, meter_store), and these are referred in elem->private_data
(it's not kctl->private_data).  And these have to be released, but
there are none doing it.  Moreover, these extra objects have to be
released only once.  Thus the release should be done only by the first
kctl element that refers to it.

For fixing these, we call either snd_usb_mixer_elem_free() (only for
kctl->private_data) or elem_private_free() (for both
kctl->private_data and elem->private_data) via kctl->private_free
appropriately.

Last but not least, snd_us16x08_controls_create() may return in the
middle without releasing the allocated *_store objects due to an
error.  For fixing this, we shuffle the allocation code so that it's
called just before its reference.

Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Reported-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-22 14:24:09 +01:00
Takashi Sakamoto 89b593c30e ALSA: usb-audio: purge needless variable length array
Variable length array is used in 'snd_us16x08_meter_get()', while there
is no need. It's better to purge it because variable length array has
overhead for stack handling.

This commit replaces the array with static length. Sparse generated below
warning.

sound/usb/mixer_us16x08.c:714:18: warning: Variable length array is used.

Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-22 07:12:23 +01:00
Bhumika Goyal 04bab35044 ALSA: usb-audio: constify snd_kcontrol_new structures
Declare snd_kcontrol_new structures as const as they are only passed as
an argument to the function add_new_ctl. This agrument is of type const,
so snd_kcontrol_new structures having this property can be made const
too.

Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-21 22:02:03 +01:00
Takashi Sakamoto 34371d236e ALSA: usb-audio: localize function without external linkage
When accessed inner a file, functions should have static qualifier for
local-linkage.

This commit fixes the bug. Sparse generated below warning.

sound/usb/mixer_us16x08.c:1043:32: warning: symbol 'snd_us16x08_create_meter_store' was not declared. Should it be static?

Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-20 21:53:14 +01:00
Takashi Sakamoto 02ed051fe1 ALSA: usb-audio: localize one-referrer variable
When accessed by one referrer inner a file, variables should have static
qualifier to declare local-linkage.

This commit fixes the bug. Sparse generated below warnings.
sound/usb/mixer_us16x08.c:156:13: warning: duplicate const
sound/usb/mixer_us16x08.c:156:18: warning: symbol 'route_names' was not declared. Should it be static?

Fixes: d2bb390a20 ("ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-20 21:45:37 +01:00
Detlef Urban d2bb390a20 ALSA: usb-audio: Tascam US-16x08 DSP mixer quirk
Add mixer quirk for Tascam US-16x08 usb interface.
Even that this is an usb compliant device,
the input channels and DSP functions (EQ/Compressor) aren't accessible
by default.

Signed-off-by: Detlef Urban <onkel@paraair.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-20 10:59:54 +01:00
Andrej Krutak 5d81296b5e ALSA: line6: Always setup isochronous transfer properties
While not all line6 devices currently support PCM, it causes no
harm to 'have it prepared'.

This also fixes toneport, which only has PCM - in which case
we previously skipped the USB transfer properties detection completely.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-07 09:40:49 +01:00
Takashi Iwai 1e94320ffe Merge branch 'for-linus' into for-next 2017-02-07 09:38:44 +01:00
Takashi Iwai f3d83317a6 Revert "ALSA: line6: Only determine control port properties if needed"
This reverts commit f6a0dd107a.

The commit caused a regression on LINE6 Transport that has no control
caps.  Although reverting the commit may result back in a spurious
error message for some device again, it's the simplest regression fix,
hence it's taken as is at first.  The further code fix will follow
later.

Fixes: f6a0dd107a ("ALSA: line6: Only determine control port properties if needed")
Reported-by: Igor Zinovev <zinigor@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-02-07 09:37:11 +01:00
Takashi Iwai e1a063f43a Merge branch 'topic/intel-lpe-audio' into for-next 2017-01-25 22:11:17 +01:00
Takashi Iwai f43e5407e4 ALSA: usb: Constify snd_rawmidi_ops
Now snd_rawmidi_ops is maintained as a const pointer in snd_rawmidi,
we can constify the definitions.

Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-01-12 12:50:48 +01:00
Dennis Kadioglu 2e40795c3b ALSA: usb-audio: Add a quirk for Plantronics BT600
Plantronics BT600 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1" and "cannot get freq at
ep 0x82". This patch adds the USB ID of the BT600 to quirks.c and
avoids those error messages.

Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-01-10 11:42:31 +01:00
Ioan-Adrian Ratiu 13a6c8328e ALSA: usb-audio: test EP_FLAG_RUNNING at urb completion
Testing EP_FLAG_RUNNING in snd_complete_urb() before running the completion
logic allows us to save a few cpu cycles by returning early, skipping the
pending urb in case the stream was stopped; the stop logic handles the urb
and sets the completion callbacks to NULL.

Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-01-05 07:35:17 +01:00
Ioan-Adrian Ratiu 1d0f953086 ALSA: usb-audio: Fix irq/process data synchronization
Commit 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream") was
incomplete causing another more severe kernel panic, so it got reverted.
This fixes both the original problem and its fallout kernel race/crash.

The original fix is to move the endpoint member NULL clearing logic inside
wait_clear_urbs() so the irq triggering the urb completion doesn't call
retire_capture/playback_urb() after the NULL clearing and generate a panic.

However this creates a new race between snd_usb_endpoint_start()'s call
to wait_clear_urbs() and the irq urb completion handler which again calls
retire_capture/playback_urb() leading to a new NULL dereference.

We keep the EP deactivation code in snd_usb_endpoint_start() because
removing it will break the EP reference counting (see [1] [2] for info),
however we don't need the "can_sleep" mechanism anymore because a new
function was introduced (snd_usb_endpoint_sync_pending_stop()) which
synchronizes pending stops and gets called inside the pcm prepare callback.

It also makes sense to remove can_sleep because it was also removed from
deactivate_urbs() signature in [3] so we benefit from more simplification.

[1] commit 015618b90 ("ALSA: snd-usb: Fix URB cancellation at stream start")
[2] commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream")
[3] commit ccc1696d5 ("ALSA: usb-audio: simplify endpoint deactivation code")

Fixes: f8114f8583 ("Revert "ALSA: usb-audio: Fix race at stopping the stream"")

Signed-off-by: Ioan-Adrian Ratiu <adi@adirat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2017-01-05 07:35:00 +01:00
Jussi Laako 3eff682d76 ALSA: usb-audio: Support both DSD LE/BE Amanero firmware versions
Add DSD support for both little endian (DSD_U32_LE) and big endian
(DSD_U32_BE) version of the Amanero firmware.

Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-28 16:07:11 +01:00
Takashi Iwai f8114f8583 Revert "ALSA: usb-audio: Fix race at stopping the stream"
This reverts commit 16200948d8.

The commit was intended to cover the race condition, but it introduced
yet another regression for devices with the implicit feedback, leading
to a kernel panic due to NULL-dereference in an irq context.

As the race condition that was addressed by the commit is very rare
and the regression is much worse, let's revert the commit for rc1, and
fix the issue properly in a later patch.

Fixes: 16200948d8 ("ALSA: usb-audio: Fix race at stopping the stream")
Reported-by: Ioan-Adrian Ratiu <adi@adirat.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2016-12-21 09:48:14 -08:00
Jussi Laako 995c6a7fd9 ALSA: hiface: Fix M2Tech hiFace driver sampling rate change
Sampling rate changes after first set one are not reflected to the
hardware, while driver and ALSA think the rate has been changed.

Fix the problem by properly stopping the interface at the beginning of
prepare call, allowing new rate to be set to the hardware. This keeps
the hardware in sync with the driver.

Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-12 22:46:43 +01:00
Nobutaka Okabe 0120073091 ALSA: usb-audio: Eliminate noise at the start of DSD playback.
[Problem]
In some USB DACs, a terrible pop noise comes to be heard
at the start of DSD playback (in the following situations).

- play first DSD track
- change from PCM track to DSD track
- change from DSD64 track to DSD128 track (and etc...)
- seek DSD track
- Fast-Forward/Rewind DSD track

[Cause]
At the start of playback, there is a little silence.
The silence bit pattern "0x69" is required on DSD mode,
but it is not like that.

[Solution]
This patch adds DSD silence pattern to the endpoint settings.

Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-12 22:43:35 +01:00
Nobutaka Okabe 7f38ca047b ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
This patch adds native DSD support for the following devices.

- TEAC NT-503
- TEAC UD-503
- TEAC UD-501

(1) Add quirks for native DSD support for TEAC devices.
(2) A specific vendor command is needed to switch between PCM/DOP and
    DSD mode, same as Denon/Marantz devices.

Signed-off-by: Nobutaka Okabe <nob77413@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-12 22:42:33 +01:00
Takashi Iwai d71bb23a81 Merge branch 'for-linus' into for-next 2016-12-09 11:21:35 +01:00
Con Kolivas 82ffb6fc63 ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks
The Logitech QuickCam Communicate Deluxe/S7500 microphone fails with the
following warning.

[    6.778995] usb 2-1.2.2.2: Warning! Unlikely big volume range (=3072),
cval->res is probably wrong.
[    6.778996] usb 2-1.2.2.2: [5] FU [Mic Capture Volume] ch = 1, val =
4608/7680/1

Adding it to the list of devices in volume_control_quirks makes it work
properly, fixing related typo.

Signed-off-by: Con Kolivas <kernel@kolivas.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-09 11:21:02 +01:00
Alberto Aguirre 17f08b0d9a ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
The Axe-Fx II implicit feedback end point and the data sync endpoint
are in different interface descriptors. Add quirk to ensure a sync
endpoint is properly configured.

Signed-off-by: Alberto Aguirre <albaguirre@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-09 11:19:31 +01:00
Andreas Pape fd1a505961 ALSA: usb-audio: more tolerant packetsize
since commit 57e6dae108 ("ALSA: usb-audio: do not trust too-big
wMaxPacketSize values"), the expected packetsize is always limited
to nominal + 25%. It was discovered, that some devices (Android audio
accessory) have a much higher jitter in used packetsizes than 25%
which would result in BABBLE condition and dropping of packets.
A better solution is so assume the jitter to be the nominal packetsize:
-one nearly empty packet followed by a almost 150% sized one.

V2: changed to assume max frequency is +50 of nominal packetsize.

Signed-off-by: Andreas Pape <apape@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-06 13:55:59 +01:00
Daniel Girnus 1e2e3fe480 ALSA: usb-audio: avoid setting of sample rate multiple times on bus
Some of userland applications call 'snd_pcm_hw_params()' and
'snd_pcm_hw_prepare()' sequentially, which means 'snd_pcm_hw_prepare()'
is called twice and the second 'snd_pcm_hw_prepare()' is called in
'SNDRV_PCM_STATE_PREPARED' state.

Some devices are not able to manage this and they will stop playback
if the sample rate will be configured several times over USB protocol.

V2: updated Changelog

Signed-off-by: Daniel Girnus <dgirnus@de.adit-jv.com>
Signed-off-by: Jens Lorenz <jlorenz@de.adit-jv.com>
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-06 13:55:15 +01:00
Takashi Iwai 16200948d8 ALSA: usb-audio: Fix race at stopping the stream
We've got a kernel crash report showing like:

  Unable to handle kernel NULL pointer dereference at virtual address 00000008 pgd = a1d7c000
  [00000008] *pgd=31c93831, *pte=00000000, *ppte=00000000
  Internal error: Oops: 17 [#1] PREEMPT SMP ARM
  CPU: 0 PID: 250 Comm: dbus-daemon Not tainted 3.14.51-03479-gf50bdf4 #1
  task: a3ae61c0 ti: a08c8000 task.ti: a08c8000
  PC is at retire_capture_urb+0x10/0x1f4 [snd_usb_audio]
  LR is at snd_complete_urb+0x140/0x1f0 [snd_usb_audio]
  pc : [<7f0eb22c>]    lr : [<7f0e57fc>]    psr: 200e0193
  sp : a08c9c98  ip : a08c9ce8  fp : a08c9ce4
  r10: 0000000a  r9 : 00000102  r8 : 94cb3000
  r7 : 94cb3000  r6 : 94d0f000  r5 : 94d0e8e8  r4 : 94d0e000
  r3 : 7f0eb21c  r2 : 00000000  r1 : 94cb3000  r0 : 00000000
  Flags: nzCv  IRQs off  FIQs on  Mode SVC_32  ISA ARM  Segment user
  Control: 10c5387d  Table: 31d7c04a  DAC: 00000015
  Process dbus-daemon (pid: 250, stack limit = 0xa08c8238)
  Stack: (0xa08c9c98 to 0xa08ca000)
  ...
  Backtrace:
  [<7f0eb21c>] (retire_capture_urb [snd_usb_audio]) from [<7f0e57fc>] (snd_complete_urb+0x140/0x1f0 [snd_usb_audio])
  [<7f0e56bc>] (snd_complete_urb [snd_usb_audio]) from [<80371118>] (__usb_hcd_giveback_urb+0x78/0xf4)
  [<803710a0>] (__usb_hcd_giveback_urb) from [<80371514>] (usb_giveback_urb_bh+0x8c/0xc0)
  [<80371488>] (usb_giveback_urb_bh) from [<80028e3c>] (tasklet_hi_action+0xc4/0x148)
  [<80028d78>] (tasklet_hi_action) from [<80028358>] (__do_softirq+0x190/0x380)
  [<800281c8>] (__do_softirq) from [<80028858>] (irq_exit+0x8c/0xfc)
  [<800287cc>] (irq_exit) from [<8000ea88>] (handle_IRQ+0x8c/0xc8)
  [<8000e9fc>] (handle_IRQ) from [<800085e8>] (gic_handle_irq+0xbc/0xf8)
  [<8000852c>] (gic_handle_irq) from [<80509044>] (__irq_svc+0x44/0x78)
  [<80508820>] (_raw_spin_unlock_irq) from [<8004b880>] (finish_task_switch+0x5c/0x100)
  [<8004b824>] (finish_task_switch) from [<805052f0>] (__schedule+0x48c/0x6d8)
  [<80504e64>] (__schedule) from [<805055d4>] (schedule+0x98/0x9c)
  [<8050553c>] (schedule) from [<800116c8>] (do_work_pending+0x30/0xd0)
  [<80011698>] (do_work_pending) from [<8000e160>] (work_pending+0xc/0x20)
  Code: e1a0c00d e92ddff0 e24cb004 e24dd024 (e5902008)
  Kernel panic - not syncing: Fatal exception in interrupt

There is a race between retire_capture_urb() and stop_endpoints().
The latter is called at stopping the stream and it sets some endpoint
fields to NULL.  But its call is asynchronous, thus the pending
complete callback might get called after these NULL clears, and it
leads the NULL dereference like the above.

The fix is to move the NULL clearance after the synchronization,
i.e. wait_clear_urbs().  This is called at prepare and hw_free
callbacks, so it's assured to be called before the restart of the
stream or the release of the stream.

Also, while we're at it, put the EP_FLAG_RUNNING flag check at the
beginning of snd_complete_urb() to skip the pending complete after the
stream is stopped.

Fixes: b2eb950de2 ("ALSA: usb-audio: stop both data and sync...")
Reported-by: Jiada Wang <jiada_wang@mentor.com>
Reported-by: Mark Craske <Mark_Craske@mentor.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-12-05 11:19:38 +01:00
Takashi Iwai 4763601a56 ALSA: usb-audio: Fix bogus error return in snd_usb_create_stream()
The function returns -EINVAL even if it builds the stream properly.
The bogus error code sneaked in during the code refactoring, but it
wasn't noticed until now since the returned error code itself is
ignored in anyway.  Kill it here, but there is no behavior change by
this patch, obviously.

Fixes: e5779998bf ('ALSA: usb-audio: refactor code')
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-11-29 22:28:40 +01:00
Andrej Krutak fc90172ba2 ALSA: line6: Claim pod x3 usb data interface
Userspace apps have to claim USB interfaces before using endpoints in
them (drivers/usb/core/devio.c:checkintf()). It's a lock mechanism so
that two "drivers" don't steal data from each other. Kernel drivers don't
have to claim interfaces to work - but they should, to lock out userspace.

While there, fix line6_properties struct to match checkpatch.pl.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-11-29 22:16:14 +01:00
Takashi Iwai 6ff1a25318 ALSA: usb-audio: Fix use-after-free of usb_device at disconnect
The usb-audio driver implements the deferred device disconnection for
the device in use.  In this mode, the disconnection callback returns
immediately while the actual ALSA card object removal happens later
when all files get closed.  As Shuah reported, this code flow,
however, leads to a use-after-free, detected by KASAN:

 BUG: KASAN: use-after-free in snd_usb_audio_free+0x134/0x160 [snd_usb_audio] at addr ffff8801c863ce10
 Write of size 8 by task pulseaudio/2244
 Call Trace:
  [<ffffffff81b31473>] dump_stack+0x67/0x94
  [<ffffffff81564ef1>] kasan_object_err+0x21/0x70
  [<ffffffff8156518a>] kasan_report_error+0x1fa/0x4e0
  [<ffffffff81564ad7>] ? kasan_slab_free+0x87/0xb0
  [<ffffffff81565733>] __asan_report_store8_noabort+0x43/0x50
  [<ffffffffa0fc0f54>] ? snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
  [<ffffffffa0fc0f54>] snd_usb_audio_free+0x134/0x160 [snd_usb_audio]
  [<ffffffffa0fc0fb1>] snd_usb_audio_dev_free+0x31/0x40 [snd_usb_audio]
  [<ffffffff8243c78a>] __snd_device_free+0x12a/0x210
  [<ffffffff8243d1f5>] snd_device_free_all+0x85/0xd0
  [<ffffffff8242cae4>] release_card_device+0x34/0x130
  [<ffffffff81ef1846>] device_release+0x76/0x1e0
  [<ffffffff81b37ad7>] kobject_release+0x107/0x370
  .....
 Object at ffff8801c863cc80, in cache kmalloc-2048 size: 2048
 Allocated:
  [<ffffffff810804eb>] save_stack_trace+0x2b/0x50
  [<ffffffff81564296>] save_stack+0x46/0xd0
  [<ffffffff8156450d>] kasan_kmalloc+0xad/0xe0
  [<ffffffff81560d1a>] kmem_cache_alloc_trace+0xfa/0x240
  [<ffffffff8214ea47>] usb_alloc_dev+0x57/0xc90
  [<ffffffff8216349d>] hub_event+0xf1d/0x35f0
  ....
 Freed:
  [<ffffffff810804eb>] save_stack_trace+0x2b/0x50
  [<ffffffff81564296>] save_stack+0x46/0xd0
  [<ffffffff81564ac1>] kasan_slab_free+0x71/0xb0
  [<ffffffff81560929>] kfree+0xd9/0x280
  [<ffffffff8214de6e>] usb_release_dev+0xde/0x110
  [<ffffffff81ef1846>] device_release+0x76/0x1e0
  ....

It's the code trying to clear drvdata of the assigned usb_device where
the usb_device itself was already released in usb_release_dev() after
the disconnect callback.

This patch fixes it by checking whether the code path is via the
disconnect callback, i.e. chip->shutdown flag is set.

Fixes: 79289e2419 ('ALSA: usb-audio: Refer to chip->usb_id for quirks...')
Reported-and-tested-by: Shuah Khan <shuahkh@osg.samsung.com>
Cc: <stable@vger.kernel.org> # v4.6+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-11-14 22:05:04 +01:00
Marcel Hasler bdc3478f90 ALSA: usb-audio: Add quirk for Syntek STK1160
The stk1160 chip needs QUIRK_AUDIO_ALIGN_TRANSFER. This patch resolves
the issue reported on the mailing list
(http://marc.info/?l=linux-sound&m=139223599126215&w=2) and also fixes
bug 180071 (https://bugzilla.kernel.org/show_bug.cgi?id=180071).

Signed-off-by: Marcel Hasler <mahasler@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-10-27 12:07:19 +02:00
Dan Carpenter fdd8218d7d ALSA: line6: fix a crash in line6_hwdep_write()
The error checking here is messed up so we could end up dereferencing
-EFAULT.

Fixes: a16039cbf1 ('ALSA: line6: Add hwdep interface to access the POD control messages')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-10-12 20:10:23 +02:00
Andrej Krutak b907900ec4 ALSA: line6: Fix POD X3 Live audio input
The commit c039aaa77a was incomplete,
missing part of the setup for Live. This makes also audio input work,
in addition to audio output.

Fixes: c039aaa77a
Reported-by: Eddi De Pieri <eddi@depieri.net>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-10-10 12:07:29 +02:00
Takashi Iwai 3383c5c395 Merge branch 'for-linus' into for-next 2016-09-29 18:20:14 +02:00
Takashi Sakamoto 8da08ca03b ALSA: usb-line6: use the same declaration as definition in header for MIDI manufacturer ID
Currently, usb-line6 module exports an array of MIDI manufacturer ID and
usb-pod module uses it. However, the declaration is not the definition in
common header. The difference is explicit length of array. Although
compiler calculates it and everything goes well, it's better to use the
same representation between definition and declaration.

This commit fills the length of array for usb-line6 module. As a small
good sub-effect, this commit suppress below warnings from static analysis
by sparse v0.5.0.

sound/usb/line6/driver.c:274:43: error: cannot size expression
sound/usb/line6/driver.c:275:16: error: cannot size expression
sound/usb/line6/driver.c:276:16: error: cannot size expression
sound/usb/line6/driver.c:277:16: error: cannot size expression

Fixes: 705ececd1c ("Staging: add line6 usb driver")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-25 22:17:45 +02:00
Anssi Hannula eb1a74b7be ALSA: usb-audio: Extend DragonFly dB scale quirk to cover other variants
The DragonFly quirk added in 42e3121d90 ("ALSA: usb-audio: Add a more
accurate volume quirk for AudioQuest DragonFly") applies a custom dB map
on the volume control when its range is reported as 0..50 (0 .. 0.2dB).

However, there exists at least one other variant (hw v1.0c, as opposed
to the tested v1.2) which reports a different non-sensical volume range
(0..53) and the custom map is therefore not applied for that device.

This results in all of the volume change appearing close to 100% on
mixer UIs that utilize the dB TLV information.

Add a fallback case where no dB TLV is reported at all if the control
range is not 0..50 but still 0..N where N <= 1000 (3.9 dB). Also
restrict the quirk to only apply to the volume control as there is also
a mute control which would match the check otherwise.

Fixes: 42e3121d90 ("ALSA: usb-audio: Add a more accurate volume quirk for AudioQuest DragonFly")
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: David W <regulars@d-dub.org.uk>
Tested-by: David W <regulars@d-dub.org.uk>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-23 08:32:18 +02:00
Valdis Kletnieks 0da325afbd ALSA: line6: snd-usb-line6 depends on CONFIG_SND_HWDEP
ERROR: "snd_hwdep_new" [sound/usb/line6/snd-usb-line6.ko] undefined!
scripts/Makefile.modpost:91: recipe for target '__modpost' failed
make[1]: *** [__modpost] Error 1

Fixes: a16039cbf1 ('ALSA: line6: Add hwdep interface to access the POD control messages')
Signed-off-by: Valdis Kletnieks <valdis.kletnieks@vt.edu>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-21 14:36:22 +02:00
kbuild test robot a4bc746c41 ALSA: line6: fix ifnullfree.cocci warnings
sound/usb/line6/driver.c:484:2-7: WARNING: NULL check before freeing functions like kfree, debugfs_remove, debugfs_remove_recursive or usb_free_urb is not needed. Maybe consider reorganizing relevant code to avoid passing NULL values.

 NULL check before some freeing functions is not needed.

 Based on checkpatch warning
 "kfree(NULL) is safe this check is probably not required"
 and kfreeaddr.cocci by Julia Lawall.

Generated by: scripts/coccinelle/free/ifnullfree.cocci

CC: Andrej Krutak <dev@andree.sk>
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-20 12:53:23 +02:00
Andrej Krutak a16039cbf1 ALSA: line6: Add hwdep interface to access the POD control messages
We must do it this way, because e.g. POD X3 won't play any sound unless
the host listens on the bulk EP, so we cannot export it only via libusb.

The driver currently doesn't use the bulk EP messages in other way,
in future it could e.g. sense/modify volume(s).

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:03:47 +02:00
Andrej Krutak cfa769695d ALSA: line6: Cleanup podhd initialization
Only initialize PCM for POD HD devices that support it.
No POD HD seems to support MIDI, thus drop the initialization.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:02:36 +02:00
Andrej Krutak f6a0dd107a ALSA: line6: Only determine control port properties if needed
Not all line6 devices use the control port.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:02:29 +02:00
Andrej Krutak c039aaa77a ALSA: line6: Add support for POD X3 Live (only USB ID differs from POD X3)
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:02:21 +02:00
Andrej Krutak 790869dacc ALSA: line6: Add support for POD X3
This includes audio in/out and basic initialization via control EP (emulates
what original driver does). The initialization is done similarly to original
POD, firmware and serial IDs are read and exported via sysfs.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:02:06 +02:00
Andrej Krutak 7811a3ad18 ALSA: line6: Allow processing of raw incoming messages
Not all PODs use MIDI via USB data interface, thus allow avoiding
that code and instead using direct processing.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:01:19 +02:00
Andrej Krutak 174e1fc0bf ALSA: line6: Distinguish device init (ctrl EP) and MIDI data transfer (int EP)
POD X3 can initialize similarly to older PODs, but it doesn't have the MIDI
interface. Instead, configuration is done via proprietary bulk EP messages.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:00:47 +02:00
Andrej Krutak f56742cc41 ALSA: line6: Add LINE6_CAP_IN_NEEDS_OUT, a void playback stream during capture
E.g. POD X3 seems to require playback data to be sent to it to generate
capture data. Otherwise the device stalls and doesn't send any more capture
data until it's reset.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:00:37 +02:00
Andrej Krutak 97d78acfb8 ALSA: line6: Allow different channel numbers for in/out
Changes bytes_per_frame to bytes_per_channel.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:00:16 +02:00
Andrej Krutak 7a0f55aeeb ALSA: line6: Support assymetrical in/out configurations
Splits max_packet_size to max_packet_size_in/out (e.g. for
different channel counts).

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 23:00:10 +02:00
Andrej Krutak 79faa2b048 ALSA: line6: Add high-speed USB support
This has two parts:
* intervals_per_second setup
  (high speed needs 8000, instead of 1000)
* iso_buffers setup (count of iso buffers depends on
  USB speed, 2 is not enough for high speed)

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 22:58:50 +02:00
Andrej Krutak b2233d97a0 ALSA: line6: Enable different number of URBs for frame transfers
This basically changes LINE6_ISO_BUFFERS constant to a configurable
iso_buffers property.

Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-09-19 22:58:08 +02:00
Takashi Iwai 3d2f4d0c0d Merge branch 'for-linus' into for-next
Back-merge from for-linus just to make the further development easier.
2016-09-11 09:33:12 +02:00
Ken Lin 83d9956b7e ALSA: usb-audio: Add sample rate inquiry quirk for B850V3 CP2114
Avoid getting sample rate on B850V3 CP2114 as it is unsupported and
causes noisy "current rate is different from the runtime rate" messages
when playback starts.

Signed-off-by: Ken Lin <ken.lin@advantech.com.tw>
Signed-off-by: Akshay Bhat <akshay.bhat@timesys.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-29 16:49:18 +02:00
Shawn Lin 1bc00f32b8 ALSA: usb-audio: rmove print for failure of kmalloc
kmalloc already print similar error once failing to alloc
enough memory, so let's remove this dump here.

Signed-off-by: Shawn Lin <shawn.lin@rock-chips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22 11:41:02 +02:00
Daniel Mack 36e1ac3cf8 ALSA: usb: fine-tune Tenor error compensation value
Users of devices affected by the Tenor feedback data error report
buffer underruns, even with the +/- 0x1.0000 quirk applied.
Compensating the error with 0xf000 instead seems to reliably fix
that issue.

See

  https://sourceforge.net/p/alsa/mailman/message/35230259/

Reported-and-tested-by: Norman Nolte <norman.nolte@gmx.net>
Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22 11:40:04 +02:00
Daniel Mack ca0dd2736a ALSA: usb: use TEAC UD-H01 quirk for more devices
The quirk seems to be necessary not only for TEAC UD-H01 devices, but to
more that are based on the Tenor 8802TL chipset. Devices built by T+A
are affected too, and they apparently all use the same USB PID:PID.

Extend the quirky handling for that device as well, and rename the
quirks flag.

Reported-and-tested-by: Thomas Gresens <T.Gresens@intershop.de>
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22 11:39:56 +02:00
Daniel Mack 9abc134167 ALSA: usb: move udh01_fb_quirk setting to quirks.c
That's a quirk, after all, so move it where to all the other quirks
live.

Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22 11:39:42 +02:00
Andrej Krutak b027d11263 ALSA: line6: Fix POD sysfs attributes segfault
The commit 02fc76f6a changed base of the sysfs attributes from device to card.
The "show" callbacks dereferenced wrong objects because of this.

Fixes: 02fc76f6a7 ('ALSA: line6: Create sysfs via snd_card_add_dev_attr()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22 11:37:31 +02:00
Andrej Krutak adc8a43a6d ALSA: line6: Give up on the lock while URBs are released.
Done, because line6_stream_stop() locks and calls line6_unlink_audio_urbs(),
which in turn invokes audio_out_callback(), which tries to lock 2nd time.

Fixes:

=============================================
[ INFO: possible recursive locking detected ]
4.4.15+ #15 Not tainted
---------------------------------------------
mplayer/3591 is trying to acquire lock:
 (&(&line6pcm->out.lock)->rlock){-.-...}, at: [<bfa27655>] audio_out_callback+0x70/0x110 [snd_usb_line6]

but task is already holding lock:
 (&(&line6pcm->out.lock)->rlock){-.-...}, at: [<bfa26aad>] line6_stream_stop+0x24/0x5c [snd_usb_line6]

other info that might help us debug this:
 Possible unsafe locking scenario:

       CPU0
       ----
  lock(&(&line6pcm->out.lock)->rlock);
  lock(&(&line6pcm->out.lock)->rlock);

 *** DEADLOCK ***

 May be due to missing lock nesting notation

3 locks held by mplayer/3591:
 #0:  (snd_pcm_link_rwlock){.-.-..}, at: [<bf8d49a7>] snd_pcm_stream_lock+0x1e/0x40 [snd_pcm]
 #1:  (&(&substream->self_group.lock)->rlock){-.-...}, at: [<bf8d49af>] snd_pcm_stream_lock+0x26/0x40 [snd_pcm]
 #2:  (&(&line6pcm->out.lock)->rlock){-.-...}, at: [<bfa26aad>] line6_stream_stop+0x24/0x5c [snd_usb_line6]

stack backtrace:
CPU: 0 PID: 3591 Comm: mplayer Not tainted 4.4.15+ #15
Hardware name: Generic AM33XX (Flattened Device Tree)
[<c0015d85>] (unwind_backtrace) from [<c001253d>] (show_stack+0x11/0x14)
[<c001253d>] (show_stack) from [<c02f1bdf>] (dump_stack+0x8b/0xac)
[<c02f1bdf>] (dump_stack) from [<c0076f43>] (__lock_acquire+0xc8b/0x1780)
[<c0076f43>] (__lock_acquire) from [<c007810d>] (lock_acquire+0x99/0x1c0)
[<c007810d>] (lock_acquire) from [<c06171e7>] (_raw_spin_lock_irqsave+0x3f/0x4c)
[<c06171e7>] (_raw_spin_lock_irqsave) from [<bfa27655>] (audio_out_callback+0x70/0x110 [snd_usb_line6])
[<bfa27655>] (audio_out_callback [snd_usb_line6]) from [<c04294db>] (__usb_hcd_giveback_urb+0x53/0xd0)
[<c04294db>] (__usb_hcd_giveback_urb) from [<c046388d>] (musb_giveback+0x3d/0x98)
[<c046388d>] (musb_giveback) from [<c04647f5>] (musb_urb_dequeue+0x6d/0x114)
[<c04647f5>] (musb_urb_dequeue) from [<c042ac11>] (usb_hcd_unlink_urb+0x39/0x98)
[<c042ac11>] (usb_hcd_unlink_urb) from [<bfa26a87>] (line6_unlink_audio_urbs+0x6a/0x6c [snd_usb_line6])
[<bfa26a87>] (line6_unlink_audio_urbs [snd_usb_line6]) from [<bfa26acb>] (line6_stream_stop+0x42/0x5c [snd_usb_line6])
[<bfa26acb>] (line6_stream_stop [snd_usb_line6]) from [<bfa26fe7>] (snd_line6_trigger+0xb6/0xf4 [snd_usb_line6])
[<bfa26fe7>] (snd_line6_trigger [snd_usb_line6]) from [<bf8d47b7>] (snd_pcm_do_stop+0x36/0x38 [snd_pcm])
[<bf8d47b7>] (snd_pcm_do_stop [snd_pcm]) from [<bf8d462f>] (snd_pcm_action_single+0x22/0x40 [snd_pcm])
[<bf8d462f>] (snd_pcm_action_single [snd_pcm]) from [<bf8d46f9>] (snd_pcm_action+0xac/0xb0 [snd_pcm])
[<bf8d46f9>] (snd_pcm_action [snd_pcm]) from [<bf8d4b61>] (snd_pcm_drop+0x38/0x64 [snd_pcm])
[<bf8d4b61>] (snd_pcm_drop [snd_pcm]) from [<bf8d6233>] (snd_pcm_common_ioctl1+0x7fe/0xbe8 [snd_pcm])
[<bf8d6233>] (snd_pcm_common_ioctl1 [snd_pcm]) from [<bf8d6779>] (snd_pcm_playback_ioctl1+0x15c/0x51c [snd_pcm])
[<bf8d6779>] (snd_pcm_playback_ioctl1 [snd_pcm]) from [<bf8d6b59>] (snd_pcm_playback_ioctl+0x20/0x28 [snd_pcm])
[<bf8d6b59>] (snd_pcm_playback_ioctl [snd_pcm]) from [<c016714b>] (do_vfs_ioctl+0x3af/0x5c8)

Fixes: 63e20df1e5 ('ALSA: line6: Reorganize PCM stream handling')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22 11:37:16 +02:00
Andrej Krutak 7e4379eae0 ALSA: line6: Remove double line6_pcm_release() after failed acquire.
If there's an error, pcm is released in line6_pcm_acquire already.

Fixes: 247d95ee6d ('ALSA: line6: Handle error from line6_pcm_acquire()')
Cc: <stable@vger.kernel.org> # v4.0+
Reviewed-by: Stefan Hajnoczi <stefanha@gmail.com>
Signed-off-by: Andrej Krutak <dev@andree.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-22 11:36:58 +02:00
Wolfram Sang 289ef67908 ALSA: usb: caiaq: audio: don't print error when allocating urb fails
kmalloc will print enough information in case of failure.

Signed-off-by: Wolfram Sang <wsa-dev@sang-engineering.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-11 23:05:57 +02:00
Vittorio Gambaletta (VittGam) 41f5e3bdbf ALSA: usb-audio: Add quirk for ELP HD USB Camera
The ELP HD USB Camera (05a3:9420) needs this 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>
2016-08-09 08:49:18 +02:00
Piotr Karasinski 7627e40c66 ALSA: usb-audio: Add a sample rate quirk for Creative Live! Cam Socialize HD (VF0610)
VF0610 does not support reading the sample rate which leads to many
lines of "cannot get freq at ep 0x82". This patch adds the USB ID
(0x041E:4080) to snd_usb_get_sample_rate_quirk() list.

Signed-off-by: Piotr Karasinski <peter.karasinski@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-08-09 08:46:40 +02:00
Takashi Iwai cf81d6b583 Merge branch 'for-next' into for-linus
Merged 4.8 changes.
2016-07-25 17:01:14 +02:00
Kazuki Oikawa 76df529697 ALSA: usb-audio: Fix quirks code is not called
snd_usb_{set_interface,ctl_msg}_quirk checks chip->usb_id to need
calling a quirks code. But existed code path that not calling
dev_set_drvdata in usb_audio_probe.

Fixes: 79289e2419 ("ALSA: usb-audio: Refer to chip->usb_id for quirks and MIDI creation")
Signed-off-by: Kazuki Oikawa <k@oikw.org>
Cc: <stable@vger.kernel.org> # v4.6+
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Tested-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-07-18 12:32:45 +02:00
Amitoj Kaur Chawla e5c5327871 ALSA: usb-audio: Change structure initialisation to C99 style
To allow for structure randomisation, replace the in order struct
initialisation style with explicit field style.

The Coccinelle semantic patch used to make this change is as follows:

@decl@
identifier i1,fld;
type T;
field list[n] fs;
@@

struct i1 {
 fs
 T fld;
 ...};

@@
identifier decl.i1,i2,decl.fld;
expression e;
position bad.p, bad.fix;
@@

struct i1 i2@p = { ...,
+ .fld = e
- e@fix
 ,...};

Signed-off-by: Amitoj Kaur Chawla <amitoj1606@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-06-17 16:58:41 +02:00
Takashi Iwai 581abbaa03 Merge branch 'for-next' into for-linus 2016-05-16 09:13:08 +02:00
Takashi Iwai 84add303ef ALSA: usb-audio: Yet another Phoneix Audio device quirk
Phoenix Audio has yet another device with another id (even a different
vendor id, 0556:0014) that requires the same quirk for the sample
rate.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-05-11 18:12:49 +02:00
Takashi Iwai 2e00fde5c6 Merge branch 'for-linus' into for-next 2016-05-10 16:06:04 +02:00
Oliver Neukum 89e448b33a ALSA: usb-midi: correct speed checking
Allow for SS+ USB devices

Signed-off-by: Oliver Neukum <oneukum@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-05-08 11:42:30 +02:00
Oliver Neukum 748a1ccc43 ALSA: usb-audio: correct speed checking
Allow handling SS+ USB devices correctly.

Signed-off-by: Oliver Neukum <oneukum@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-05-08 11:42:04 +02:00
Takashi Iwai 2d2c038a99 ALSA: usb-audio: Quirk for yet another Phoenix Audio devices (v2)
Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the
same workaround as TMX320 for avoiding the firmware bug.  It fixes the
frequent error about the sample rate inquiries and the slow device
probe as consequence.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-29 13:40:25 +02:00
Takashi Iwai 57dd5414a0 ALSA: usb-audio: Limit retrying sample rate reads
There are many USB audio devices with buggy firmware that don't react
with the sample rate reading properly.  This often results in the
flood of error messages and slowing down the operation.

The sample rate read back is basically only for confirming the sample
rate setup, and it's not critically important.  As a compromise, in
this patch, we stop the sample rate read back once when the device
gives errors more than tolerance (twice, as of now).  This should
improve most of error cases while we still can catch the firmware
bugginess.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-29 11:49:04 +02:00
Takashi Iwai a33d595996 Merge branch 'for-linus' into for-next
For taking back the recent change of HDA HDMI fixes for i915 HSW/BDW.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-26 10:12:46 +02:00
Linus Torvalds c8c52850e6 sound fixes for 4.6-rc4
we've had a very calm development cycle, so far.  Here are the few
 fixes for HD-audio and USB-audio, all of which are small and easy.
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Merge tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "We've had a very calm development cycle, so far.  Here are the few
  fixes for HD-audio and USB-audio, all of which are small and easy"

* tag 'sound-4.6-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix inconsistent monitor_present state until repoll
  ALSA: hda - Fix regression of monitor_present flag in eld proc file
  ALSA: usb-audio: Skip volume controls triggers hangup on Dell USB Dock
  ALSA: hda/realtek - Enable the ALC292 dock fixup on the Thinkpad T460s
  ALSA: sscape: Use correct format identifier for size_t
  ALSA: usb-audio: Add a quirk for Plantronics BT300
  ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320
  ALSA: hda - Bind with i915 only when Intel graphics is present
2016-04-14 18:47:51 -07:00
Kailang Yang adcdd0d5a1 ALSA: usb-audio: Skip volume controls triggers hangup on Dell USB Dock
This is Dell usb dock audio workaround.
It was fixed the master volume keep lower.

[Some background: the patch essentially skips the controls of a couple
 of FU volumes.  Although the firmware exposes the dB and the value
 information via the usb descriptor, changing the values (we set the
 min volume as default) screws up the device.  Although this has been
 fixed in the newer firmware, the devices are shipped with the old
 firmware, thus we need the workaround in the driver side.  -- tiwai]

Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-12 11:24:15 +02:00
Daniel Mack 191227d99a ALSA: usb-audio: allow clock source validity interrupts
miniDSP USBStreamer UAC2 devices send clock validity changes with the
control field set to zero. The current interrupt handler ignores all
packets if the control field does not match the mixer element's, but
it really should only do that in case that field is needed to
distinguish multiple elements with the same ID.

This patch implements a logic that lets notifications packets pass
if the element ID is unique for a given device.

Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-09 17:22:40 +02:00
Daniel Mack cddaafb9a4 ALSA: usb-audio: add UAC2 clock sources as mixer controls
UAC2 specifies clock sources that optionally have validity controls.
This patch exposes them as mixer controls, so they can be read (and
at least in theory even be written) by userspace applications in order
to make clock selection policy decisions.

This implementation does nothing if the device is not UAC2 compliant,
or if the clock source does not define said validity control bits.

Tested with a miniDSP USBStreamer (0x2752/0x0016).

Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-09 17:22:11 +02:00
Dennis Kadioglu b4203ff546 ALSA: usb-audio: Add a quirk for Plantronics BT300
Plantronics BT300 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x1". This patch adds the USB
ID of the BT300 to quirks.c and avoids those error messages.

Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-06 10:33:52 +02:00
Linus Torvalds 62d2def9a5 media fixes for v4.6-rc2
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Merge tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media

Pull media fixes from Mauro Carvalho Chehab:
 "Some bug fixes on au0828 and snd-usb-audio:

   - the au0828+snd-usb-audio MC patch broke several things and produced
     some race conditions.  Better to revert the patches, and re-work on
     them for a next version

   - fix a regression at tuner disable links logic

   - properly handle dev_state as a bitmask"

* tag 'media/v4.6-3' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-media:
  [media] Revert "[media] media: au0828 change to use Managed Media Controller API"
  [media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
  [media] au0828: Fix dev_state handling
  [media] au0828: fix au0828_v4l2_close() dev_state race condition
  [media] media: au0828 fix to clear enable/disable/change source handlers
  [media] v4l2-mc: cleanup a warning
  [media] au0828: disable tuner links and cache tuner/decoder
2016-04-05 06:47:50 -07:00
Takashi Iwai f03b24a851 ALSA: usb-audio: Add a sample rate quirk for Phoenix Audio TMX320
Phoenix Audio TMX320 gives the similar error when the sample rate is
asked:
  usb 2-1.3: 2:1: cannot get freq at ep 0x85
  usb 2-1.3: 1:1: cannot get freq at ep 0x2
  ....

Add the corresponding USB-device ID (1de7:0014) to
snd_usb_get_sample_rate_quirk() list.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-04-04 11:47:50 +02:00
Linus Torvalds 2708d17d07 sound fixes for 4.6-rc2
A collection of small fixes:
 - A fix in ALSA timer core to avoid possible BUG() trigger
 - A fix in ALSA timer core 32bit compat layer
 - A few HD-audio quirks for ASUS and HP machines
 - AMD HD-audio HDMI controller quirks
 - Fixes of USB-audio double-free at some error paths
 - A fix for memory leak in DICE driver at hotunplug
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Merge tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes:

   - a fix in ALSA timer core to avoid possible BUG() trigger
   - a fix in ALSA timer core 32bit compat layer
   - a few HD-audio quirks for ASUS and HP machines
   - AMD HD-audio HDMI controller quirks
   - fixes of USB-audio double-free at some error paths
   - a fix for memory leak in DICE driver at hotunplug"

* tag 'sound-4.6-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: timer: Use mod_timer() for rearming the system timer
  ALSA: hda - fix front mic problem for a HP desktop
  ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call
  ALSA: hda: add AMD Polaris-10/11 AZ PCI IDs with proper driver caps
  ALSA: dice: fix memory leak when unplugging
  ALSA: hda - Apply fix for white noise on Asus N550JV, too
  ALSA: hda - Fix white noise on Asus N750JV headphone
  ALSA: hda - Asus N750JV external subwoofer fixup
  ALSA: timer: fix gparams ioctl compatibility for different architectures
2016-04-01 17:13:23 -05:00
Mauro Carvalho Chehab c89178f57a [media] Revert "[media] sound/usb: Use Media Controller API to share media resources"
Unfortunately, this patch caused several regressions at au0828 and
snd-usb-audio, like this one:
	https://bugzilla.kernel.org/show_bug.cgi?id=115561

It also showed several troubles at the MC core that handles pretty
poorly the memory protections and data lifetime management.

So, better to revert it and fix the core before reapplying this
change.

This reverts commit aebb2b89bf ("[media] sound/usb: Use Media
Controller API to share media resources")'

Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
2016-03-31 15:02:33 -03:00
Vladis Dronov 836b34a935 ALSA: usb-audio: Fix double-free in error paths after snd_usb_add_audio_stream() call
create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.

This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.

Based on a patch by Takashi Iwai <tiwai@suse.de>

[Note for stable backports:
 this patch requires the commit 902eb7fd1e ('ALSA: usb-audio: Minor
 code cleanup in create_fixed_stream_quirk()')]

Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358
Reported-by: Ralf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: Vladis Dronov <vdronov@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-31 18:07:31 +02:00
Linus Torvalds 87cf815b9e sound fixes for 4.6-rc1
The previous pull request introduced a few WARN_ON() for Intel
 HD-audio HDMI.  Indeed it caught bugs, and now users get annoyed.
 So this request came up: a collection of small fixes to paper over
 the inconsistencies on (mostly) old Intel chipsets.
 
 In addition, a trivial USB-audio quirk is included, too.
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Merge tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "The previous pull request introduced a few WARN_ON() for Intel
  HD-audio HDMI.  Indeed it caught bugs, and now users get annoyed.  So
  this request came up: a collection of small fixes to paper over the
  inconsistencies on (mostly) old Intel chipsets.

  In addition, a trivial USB-audio quirk is included, too"

* tag 'sound-fix-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix missing ELD update at unplugging
  ALSA: usb-audio: add Microsoft HD-5001 to quirks
  ALSA: hda - Workaround for unbalanced i915 power refcount by concurrent probe
  ALSA: hda - Fix spurious kernel WARNING on Baytrail HDMI
  ALSA: hda - Fix forgotten HDMI monitor_present update
  ALSA: hda - Really restrict i915 notifier to HSW+
2016-03-22 12:13:53 -07:00
Victor Clément 0ef21100ae ALSA: usb-audio: add Microsoft HD-5001 to quirks
The Microsoft HD-5001 webcam microphone does not support sample rate
reading as the HD-5000 one.
This results in dmesg errors and sound hanging with pulseaudio.

Signed-off-by: Victor Clément <victor.clement@openmailbox.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-20 09:03:49 +01:00
Linus Torvalds 021f163d69 sound updates for 4.6-rc1
After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
 changes in the core at this time while a lot of changes are found in
 the driver side, unsurprisingly.  Below are some highlights:
 
 ALSA core:
 - A few more hardening in ALSA timer codes
 - An extension of sequencer API for advertising the card / pid
 - Small fixes in compress-offload and jack layers
 
 HD-audio:
 - Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
   DP-MST support
 - Lots of code refactoring for sharing with ASoC SKL driver
 - Regression fixes for Intel HDMI/DP
 - Fixups for CX20724 codec, Lenovo AiO
 
 USB-audio:
 - Add quirk_alias option to make quirk debugging easier
 - Fixes for possible Oops by malformed firmware
 
 Firewire:
 - Add support for FW-1804 in tascam driver
 - Improvements / changes in card registration, multi stream handling,
   etc for DICE
 - Lots of code refactoring
 
 ASoC:
 - Enhancements of still ongoing topology API
 - Lots of commits for Intel Skylake support including HDMI support
 - A few Intel Atom driver updates for recent devices
 - Lots of improvements to the Renesas drivers
 - Capture support for Qualcomm drivers
 - Support for TI DaVinci DRA7xxx devices
 - New machine drivers for Freescale systems with Cirrus CODECs,
   Mediatek systems with RT5650 CODECs
 - New CPU drivers for Allwinner S/PDIF controllers
 - New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514
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Merge tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "After a heavy storm by syzkaller in 4.5 cycle, we have relatively few
  changes in the core at this time while a lot of changes are found in
  the driver side, unsurprisingly.  Below are some highlights:

  ALSA core:
   - A few more hardening in ALSA timer codes
   - An extension of sequencer API for advertising the card / pid
   - Small fixes in compress-offload and jack layers

  HD-audio:
   - Dynamic PCM assignment in HDMI/DP codec; preparation for upcoming
     DP-MST support
   - Lots of code refactoring for sharing with ASoC SKL driver
   - Regression fixes for Intel HDMI/DP
   - Fixups for CX20724 codec, Lenovo AiO

  USB-audio:
   - Add quirk_alias option to make quirk debugging easier
   - Fixes for possible Oops by malformed firmware

  Firewire:
   - Add support for FW-1804 in tascam driver
   - Improvements / changes in card registration, multi stream handling,
     etc for DICE
   - Lots of code refactoring

  ASoC:
   - Enhancements of still ongoing topology API
   - Lots of commits for Intel Skylake support including HDMI support
   - A few Intel Atom driver updates for recent devices
   - Lots of improvements to the Renesas drivers
   - Capture support for Qualcomm drivers
   - Support for TI DaVinci DRA7xxx devices
   - New machine drivers for Freescale systems with Cirrus CODECs,
     Mediatek systems with RT5650 CODECs
   - New CPU drivers for Allwinner S/PDIF controllers
   - New CODEC drivers for Maxim MAX9867 and MAX98926 and Realtek RT5514"

* tag 'sound-4.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (291 commits)
  ALSA: hda - Fix mutex deadlock at HDMI/DP hotplug
  ALSA: ctl: change return value in compatibility layer so that it's the same value in core implementation
  ALSA: mixart: silence an uninitialized variable warning
  ALSA: usb-audio: Add sanity checks for endpoint accesses
  ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
  ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
  ALSA: hda - Limit i915 HDMI binding only for HSW and later
  ALSA: hda - Fix unconditional GPIO toggle via automute
  ALSA: mixart: silence unitialized variable warnings
  ALSA: hda - Fixes double fault in nvhdmi_chmap_cea_alloc_validate_get_type
  ALSA: intel8x0: Add clock quirk entry for AD1981B on IBM ThinkPad X41.
  ALSA: hda - Add new GPU codec ID 0x10de0082 to snd-hda
  ASoC: rsnd: add simplified module explanation
  ASoC: hdac_hdmi: Add broxton device ID
  ASoC: Intel: Bxtn: Add Broxton PCI ID
  ASoC: Intel: Skylake: Move Skylake dsp ops & loader ops
  ASoC: Intel: add dmabuffer to common sst_dsp
  ASoC: Intel: Skylake: Unstatify skl_dsp_enable_core
  ASoC: Intel: Skylake: Fix whitepsace issues
  ASoC: Intel: Skylake: Move module id defines
  ...
2016-03-18 10:05:46 -07:00
Takashi Iwai 447d6275f0 ALSA: usb-audio: Add sanity checks for endpoint accesses
Add some sanity check codes before actually accessing the endpoint via
get_endpoint() in order to avoid the invalid access through a
malformed USB descriptor.  Mostly just checking bNumEndpoints, but in
one place (snd_microii_spdif_default_get()), the validity of iface and
altsetting index is checked as well.

Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-16 12:45:32 +01:00
Takashi Iwai 902eb7fd1e ALSA: usb-audio: Minor code cleanup in create_fixed_stream_quirk()
Just a minor code cleanup: unify the error paths.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-16 12:43:27 +01:00
Takashi Iwai 0f886ca127 ALSA: usb-audio: Fix NULL dereference in create_fixed_stream_quirk()
create_fixed_stream_quirk() may cause a NULL-pointer dereference by
accessing the non-existing endpoint when a USB device with a malformed
USB descriptor is used.

This patch avoids it simply by adding a sanity check of bNumEndpoints
before the accesses.

Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=971125
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-16 12:42:16 +01:00
Mauro Carvalho Chehab 8331c055b2 Merge commit '840f5b0572ea' into v4l_for_linus
* commit '840f5b0572ea': (381 commits)
  media: au0828 disable tuner to demod link in au0828_media_device_register()
  [media] touptek: cast char types on %x printk
  [media] touptek: don't DMA at the stack
  [media] mceusb: use %*ph for small buffer dumps
  [media] v4l: exynos4-is: Drop unneeded check when setting up fimc-lite links
  [media] v4l: vsp1: Check if an entity is a subdev with the right function
  [media] hide unused functions for !MEDIA_CONTROLLER
  [media] em28xx: fix Terratec Grabby AC97 codec detection
  [media] media: add prefixes to interface types
  [media] media: rc: nuvoton: switch attribute wakeup_data to text
  [media] v4l2-ioctl: fix YUV422P pixel format description
  [media] media: fix null pointer dereference in v4l_vb2q_enable_media_source()
  [media] v4l2-mc.h: fix yet more compiler errors
  [media] staging/media: add missing TODO files
  [media] media.h: always start with 1 for the audio entities
  [media] sound/usb: Use meaninful names for goto labels
  [media] v4l2-mc.h: fix compiler warnings
  [media] media: au0828 audio mixer isn't connected to decoder
  [media] sound/usb: Use Media Controller API to share media resources
  [media] dw2102: add support for TeVii S662
  ...
2016-03-15 07:48:28 -03:00
Takashi Iwai 6defb60ae4 Merge branch 'for-linus' into for-next
Resolved the conflicts with the latest HDA HDMI fixes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-04 08:41:41 +01:00
Shuah Khan c0fd9cdf94 [media] sound/usb: Use meaninful names for goto labels
Fix to use meaningful names instead of numbered goto labels

Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com>
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
2016-03-03 18:08:13 -03:00
Shuah Khan aebb2b89bf [media] sound/usb: Use Media Controller API to share media resources
Change ALSA driver to use Media Controller API to share media resources
with DVB and V4L2 drivers on a AU0828 media device. Media Controller
specific initialization is done after sound card is registered. ALSA
creates Media interface and entity function graph nodes for Control,
Mixer, PCM Playback, and PCM Capture devices.

snd_usb_hw_params() will call Media Controller enable source handler
interface to request the media resource. If resource request is
granted, it will release it from snd_usb_hw_free(). If resource is
busy, -EBUSY is returned.

Media specific cleanup is done in usb_audio_disconnect().

Signed-off-by: Shuah Khan <shuahkh@osg.samsung.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@osg.samsung.com>
2016-03-03 15:01:13 -03:00
Dennis Kadioglu 17e2df4613 ALSA: usb-audio: Add a quirk for Plantronics DA45
Plantronics DA45 does not support reading the sample rate which leads
to many lines of "cannot get freq at ep 0x4" and "cannot get freq at
ep 0x84". This patch adds the USB ID of the DA45 to quirks.c and
avoids those error messages.

Signed-off-by: Dennis Kadioglu <denk@post.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-03-01 14:40:52 +01:00
Takashi Iwai d61b04f801 Merge branch 'for-linus' into for-next 2016-02-26 20:26:09 +01:00
Andrey Konovalov 07d86ca93d ALSA: usb-audio: avoid freeing umidi object twice
The 'umidi' object will be free'd on the error path by snd_usbmidi_free()
when tearing down the rawmidi interface. So we shouldn't try to free it
in snd_usbmidi_create() after having registered the rawmidi interface.

Found by KASAN.

Signed-off-by: Andrey Konovalov <andreyknvl@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-02-13 09:30:58 +01:00
Takashi Iwai c9e9daccc7 Merge branch 'topic/core-fixes' into for-next 2016-02-08 08:16:55 +01:00
Lev Lybin 1b3c993a69 ALSA: usb-audio: Add quirk for Microsoft LifeCam HD-6000
Microsoft LifeCam HD-6000 (045e:076f) requires the similar quirk for
avoiding the stall due to the invalid sample rate reads.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=111491
Signed-off-by: Lev Lybin <lev.lybin@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-29 17:25:39 +01:00
Jurgen Kramer ad678b4ccd ALSA: usb-audio: Add native DSD support for PS Audio NuWave DAC
This patch adds native DSD support for the PS Audio NuWave DAC.

Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-29 15:32:23 +01:00
Jurgen Kramer 5327d6ba97 ALSA: usb-audio: Fix OPPO HA-1 vendor ID
In my patch adding native DSD support for the Oppo HA-1, the wrong vendor ID got
through. This patch fixes the vendor ID and aligns the comment.

Fixes: a4eae3a506 ('ALSA: usb: Add native DSD support for Oppo HA-1')
Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-29 15:31:17 +01:00
Takashi Iwai e270336331 ALSA: usb-audio: Add quirk_alias option
This patch adds a new option "quirk_alias" to snd-usb-audio driver for
allowing user to pass the quirk alias list.  A quirk alias consists of
a string form like 0123abcd:5678beef, which makes to apply a quirk to
a device with USB ID 0123:abcd treated as if it were 5678:beef.
This feature is useful to test an existing quirk, typically for a
newer model of the same vendor, without patching / rebuilding the
kernel driver.

The current implementation is fairly simplistic: since there is no API
for matching a usb_device_id to the given ID pair, it has an open code
to loop over the id table and matches only with vendor:product pair.
So far, this is OK, as all existing entries are with vendor:product
pairs, indeed.  Once when we have another matching entry, however,
we'd need to update get_alias_quirk() as well.

Note that this option is provided only for testing / development.  If
you want to have a proper support, contact to upstream for adding the
matching quirk in the driver code statically.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-29 07:36:10 +01:00
Takashi Iwai 79289e2419 ALSA: usb-audio: Refer to chip->usb_id for quirks and MIDI creation
This is a preliminary patch for the later change to allow a better
quirk ID management.  In the current USB-audio code, there are a few
places looking at usb_device idVendor and idProduct fields directly
even though we have already a static member in snd_usb_audio.usb_id.
This patch modifies such codes to refer to the latter field.

For achieving this, two slightly intensive changes have been done:
- The snd_usb_audio object is set/reset via dev_getdrv() for the given
  USB device; it's needed for minimizing the changes for some existing
  quirks that take only usb_device object.

- __snd_usbmidi_create() is introduced to receive the pre-given usb_id
  argument.  The exported snd_usbmidi_create() is unchanged by calling
  this new function internally.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-29 07:36:10 +01:00
Guillaume Fougnies 5a4ff9ec8d ALSA: usb-audio: Fix TEAC UD-501/UD-503/NT-503 usb delay
TEAC UD-501/UD-503/NT-503 fail to switch properly between different
rate/format. Similar to 'Playback Design', this patch corrects the
invalid clock source error for TEAC products and avoids complete
freeze of the usb interface of 503 series.

Signed-off-by: Guillaume Fougnies <guillaume@eulerian.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-26 06:58:57 +01:00
Linus Torvalds a016af2e70 sound updates for 4.5-rc1
We've had quite busy weeks in this cycle.  Looking at ALSA core, the
 significant changes are a few fixes wrt timer and sequencer ioctls
 that have been revealed by fuzzer recently.  Other than that, ASoC
 core got a few updates about DAI link handling, but these are rather
 straightforward refactoring.
 
 In drivers scene, ASoC received quite lots of new drivers in addition
 to bunch of updates for still ongoing Intel Skylake support and
 topology API.  HD-audio gained a new HDMI/DP hotplug notification via
 component.  FireWire got a pile of code refactoring/updates with
 SCS.1x driver integration.
 
 More highlights are shown below.
 
 [NOTE: this contains also many commits for DRM.  This is due to the
  pull of drm stable branch into sound tree, as the base of i915 audio
  component work for HD-audio.  The highlights below don't contain
  these DRM changes, as these are supposed to be pulled via drm tree in
  anyway sooner or later.]
 
 Core
  - Handful fixes to harden ALSA timer and sequencer ioctls against
    races reported by syzkaller fuzzer
  - Irq description string can be unique to each card; only for
    HD-audio for now
 
 ASoC
  - Conversion of the array of DAI links to a list for supporting
    dynamically adding and removing DAI links
  - Topology API enhancements to make everything more component based
    and being able to specify PCM links via topology
  - Some more fixes for the topology code, though it is still not final
    and ready for enabling in production; we really need to get to the
    point where that can be done
  - A pile of changes for Intel SkyLake drivers which hopefully deliver
    some useful initial functionality for systems with this chipset,
    though there is more work still to come
  - Lots of new features and cleanups for the Renesas drivers
  - ANC support for WM5110
  - New drivers: Imagination Technologies IPs, Atmel class D speaker,
    Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
    RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
  - Rename PCM1792a driver to be generic pcm179x
 
 HD-Audio
  - Use audio component for i915 HDMI/DP hotplug handling
  - On-demand binding with i915 driver
  - bdl_pos_adj parameter adjustment for Baytrail controllers
  - Enable power_save_node for CX20722; this shouldn't lead to
    regression, hopefully
  - Kabylake HDMI/DP codec support
  - Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
    machines
  - A few code refactoring
 
 FireWire
  - Lots of code cleanup and refactoring
  - Integrate the support of SCS.1x devices into snd-oxfw driver;
    snd-scs1x driver is obsoleted
 
 USB-audio
  - Fix possible NULL dereference at disconnection
  - A regression fix for Native Instruments devices
 
 Misc
  - A few code cleanups of fm801 driver
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Merge tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We've had quite busy weeks in this cycle.  Looking at ALSA core, the
  significant changes are a few fixes wrt timer and sequencer ioctls
  that have been revealed by fuzzer recently.  Other than that, ASoC
  core got a few updates about DAI link handling, but these are rather
  straightforward refactoring.

  In drivers scene, ASoC received quite lots of new drivers in addition
  to bunch of updates for still ongoing Intel Skylake support and
  topology API.  HD-audio gained a new HDMI/DP hotplug notification via
  component.  FireWire got a pile of code refactoring/updates with
  SCS.1x driver integration.

  More highlights are shown below.

  [ NOTE: this contains also many commits for DRM.  This is due to the
    pull of drm stable branch into sound tree, as the base of i915 audio
    component work for HD-audio.  The highlights below don't contain
    these DRM changes, as these are supposed to be pulled via drm tree
    in anyway sooner or later.  ]

  Core:
   - Handful fixes to harden ALSA timer and sequencer ioctls against
     races reported by syzkaller fuzzer
   - Irq description string can be unique to each card; only for
     HD-audio for now

  ASoC:
   - Conversion of the array of DAI links to a list for supporting
     dynamically adding and removing DAI links
   - Topology API enhancements to make everything more component based
     and being able to specify PCM links via topology
   - Some more fixes for the topology code, though it is still not final
     and ready for enabling in production; we really need to get to the
     point where that can be done
   - A pile of changes for Intel SkyLake drivers which hopefully deliver
     some useful initial functionality for systems with this chipset,
     though there is more work still to come
   - Lots of new features and cleanups for the Renesas drivers
   - ANC support for WM5110
   - New drivers: Imagination Technologies IPs, Atmel class D speaker,
     Cirrus CS47L24 and WM1831, Dialog DA7128, Realtek RT5659 and
     RT56156, Rockchip RK3036, TI PC3168A, and AMD ACP
   - Rename PCM1792a driver to be generic pcm179x

  HD-Audio:
   - Use audio component for i915 HDMI/DP hotplug handling
   - On-demand binding with i915 driver
   - bdl_pos_adj parameter adjustment for Baytrail controllers
   - Enable power_save_node for CX20722; this shouldn't lead to
     regression, hopefully
   - Kabylake HDMI/DP codec support
   - Quirks for Lenovo E50-80, Dell Latitude E-series, and other Dell
     machines
   - A few code refactoring

  FireWire:
   - Lots of code cleanup and refactoring
   - Integrate the support of SCS.1x devices into snd-oxfw driver;
     snd-scs1x driver is obsoleted

  USB-audio:
   - Fix possible NULL dereference at disconnection
   - A regression fix for Native Instruments devices

  Misc:
   - A few code cleanups of fm801 driver"

* tag 'sound-4.5-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (722 commits)
  ALSA: timer: Code cleanup
  ALSA: timer: Harden slave timer list handling
  ALSA: hda - Add fixup for Dell Latitidue E6540
  ALSA: timer: Fix race among timer ioctls
  ALSA: hda - add codec support for Kabylake display audio codec
  ALSA: timer: Fix double unlink of active_list
  ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
  ALSA: hda - fix the headset mic detection problem for a Dell laptop
  ALSA: hda - Fix white noise on Dell Latitude E5550
  ALSA: hda_intel: add card number to irq description
  ALSA: seq: Fix race at timer setup and close
  ALSA: seq: Fix missing NULL check at remove_events ioctl
  ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
  ASoC: hdac_hdmi: remove unused hdac_hdmi_query_pin_connlist
  ASoC: AMD: Add missing include file
  ALSA: hda - Fixup inverted internal mic for Lenovo E50-80
  ALSA: usb: Add native DSD support for Oppo HA-1
  ASoC: Make aux_dev more like a generic component
  ASoC: bcm2835: cleanup includes by ordering them alphabetically
  ASoC: AMD: Manage ACP 2.x SRAM banks power
  ...
2016-01-17 12:05:31 -08:00
Linus Torvalds 7d1fc01afc Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
Pull trivial tree updates from Jiri Kosina.

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial:
  floppy: make local variable non-static
  exynos: fixes an incorrect header guard
  dt-bindings: fixes some incorrect header guards
  cpufreq-dt: correct dead link in documentation
  cpufreq: ARM big LITTLE: correct dead link in documentation
  treewide: Fix typos in printk
  Documentation: filesystem: Fix typo in fs/eventfd.c
  fs/super.c: use && instead of & for warn_on condition
  Documentation: fix sysfs-ptp
  lib: scatterlist: fix Kconfig description
2016-01-14 17:04:19 -08:00
Takashi Iwai c4a359a004 ALSA: usb-audio: Fix mixer ctl regression of Native Instrument devices
The commit [da6d276957ea: ALSA: usb-audio: Add resume support for
Native Instruments controls] brought a regression where the Native
Instrument audio devices don't get the correct value at update due to
the missing shift at writing.  This patch addresses it.

Fixes: da6d276957 ('ALSA: usb-audio: Add resume support for Native Instruments controls')
Reported-and-tested-by: Owen Williams <owilliams@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-13 07:24:07 +01:00
Takashi Iwai 5c06d68bc2 ALSA: usb-audio: Avoid calling usb_autopm_put_interface() at disconnect
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: Takashi Iwai <tiwai@suse.de>
2016-01-12 14:12:38 +01:00
Jurgen Kramer a4eae3a506 ALSA: usb: Add native DSD support for Oppo HA-1
This patch adds native DSD support for the Oppo HA-1. It uses a XMOS chipset
but they use their own vendor ID.

Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2016-01-11 09:55:58 +01:00
Takashi Iwai 59c8231089 Merge branch 'for-linus' into for-next
Conflicts:
	drivers/gpu/drm/i915/intel_pm.c
2015-12-23 08:33:34 +01:00
Geliang Tang f67d71ae8b ALSA: usb-audio: use list_for_each_entry_continue_reverse
For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().

Signed-off-by: Geliang Tang <geliangtang@163.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-12-22 10:58:28 +01:00
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
Julia Lawall 17074c1a5f ALSA: usb-audio: constify usb_protocol_ops structures
The usb_protocol_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>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-12-11 16:18:02 +01:00
Masanari Iida e3d132d123 treewide: Fix typos in printk
This patch fix multiple spelling typos found in
various part of kernel.

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Acked-by: Randy Dunlap <rdunlap@infradead.org>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2015-12-08 14:59:19 +01:00
Colin Ian King 82bd59bcb3 ALSA: usx2y: fix inconsistent indenting on if statement
minor change, indenting is one tab out.

Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-12-02 18:03:24 +01:00
Julia Lawall efdbe3c3ed ALSA: midi: constify snd_rawmidi_global_ops structures
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>
2015-11-22 09:21:16 +01:00
Cheah Kok Cheong 3c7a093587 ALSA: ua101: replace le16_to_cpu() with usb_endpoint_maxp()
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>
2015-11-16 09:03:06 +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