Commit Graph

80 Commits

Author SHA1 Message Date
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
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
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
Takashi Iwai
d71bb23a81 Merge branch 'for-linus' into for-next 2016-12-09 11:21:35 +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
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
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
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
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
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
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
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
1fb8510cdb ALSA: pcm: Add snd_pcm_stop_xrun() helper
Add a new helper function snd_pcm_stop_xrun() to the standard sequnce
lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the
existing open codes with this helper.

The function checks the PCM running state to prevent setting the wrong
state, too, for more safety.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-09 18:20:40 +01:00
Takashi Iwai
67e225009b ALSA: usb-audio: Trigger PCM XRUN at XRUN
The usb-audio driver detects XRUN at its complete callback, but the
actual code to trigger PCM XRUN is commented out because it caused
deadlock in the past.  This patch revives the PCM trigger properly.
It resulted in more than just enabling snd_pcm_stop(), but it had to
deduce the PCM substream with proper NULL checks and holds the stream
lock around the call.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-06 13:04:49 +01:00
Takashi Iwai
a6cece9d81 ALSA: usb-audio: Pass direct struct pointer instead of list_head
Some functions in mixer.c and endpoint.c receive list_head instead of
the object itself.  This is not obvious and rather error-prone.  Let's
pass the proper object directly instead.

The functions in midi.c still receive list_head and this can't be
changed since the object definition isn't exposed to the outside of
midi.c, so left as is.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-04 15:09:10 +01:00
Takashi Iwai
92a586bdc0 ALSA: usb-audio: Fix races at disconnection and PCM closing
When a USB-audio device is disconnected while PCM is still running, we
still see some race: the disconnect callback calls
snd_usb_endpoint_free() that calls release_urbs() and then kfree()
while a PCM stream would be closed at the same time and calls
stop_endpoints() that leads to wait_clear_urbs().  That is, the EP
object might be deallocated while a PCM stream is syncing with
wait_clear_urbs() with the same EP.

Basically calling multiple wait_clear_urbs() would work fine, also
calling wait_clear_urbs() and release_urbs() would work, too, as
wait_clear_urbs() just reads some fields in ep.  The problem is the
succeeding kfree() in snd_pcm_endpoint_free().

This patch moves out the EP deallocation into the later point, the
destructor callback.  At this stage, all PCMs must have been already
closed, so it's safe to free the objects.

Reported-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-26 10:33:35 +02:00
Clemens Ladisch
7040b6d1fe ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus
causing the PC to send the samples at a wrong rate, which results in
clicks and crackles in the output.

Add a workaround to detect and fix the corruption.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
[mick37@gmx.de: use sender->udh01_fb_quirk rather than
 ep->udh01_fb_quirk in snd_usb_handle_sync_urb()]
Reported-and-tested-by: Mick <mick37@gmx.de>
Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02 18:21:55 +02:00
Takashi Iwai
0ba41d917e ALSA: usb-audio: Use standard printk helpers
Convert with dev_err() and co from snd_printk(), etc.
As there are too deep indirections (e.g. ep->chip->dev->dev),
a few new local macros, usb_audio_err() & co, are introduced.

Also, the device numbers in some messages are dropped, as they are
shown in the prefix automatically.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-02-26 16:45:34 +01:00
Thomas Pugliese
a93455e1c3 ALSA: usb: use multiple packets per urb for Wireless USB inbound audio
For Wireless USB audio devices, use multiple isoc packets per URB for
inbound endpoints with a datainterval < 5.  This allows the WUSB host
controller to take advantage of bursting to service endpoints whose
logical polling interval is less than the 4ms minimum polling interval
limit in WUSB.

Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-27 11:55:13 +01:00
Eldad Zack
05c79b772f ALSA: usb-audio: remove unused endpoint flag EP_FLAG_ACTIVATED
EP_FLAG_ACTIVATED is never tested for, remove it.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 11:22:43 +02:00
Eldad Zack
df23a2466a ALSA: usb-audio: rename alt_idx to altsetting
As Clemens Ladisch kindly explained:
 "Please note that there are two methods to identify alternate settings:
  the number, which is the value in bAlternateSetting, and the index,
  which is the index in the descriptor array.  There might be some wording
  in the USB spec that these two values must be the same, but in reality,
  [insert standard rant about firmware writers], bAlternateSetting
  must be treated as a random ID value."

This patch changes the name to express the correct usage semantics.
No functional change.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 11:22:03 +02:00
Eldad Zack
9b7c552bba ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate()
The return value of snd_usb_endpoint_deactivate() is not used,
make the function have no return value.
Update the documentation to reflect what the function is actually
doing.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 11:00:03 +02:00
Eldad Zack
239b9f7990 ALSA: usb-audio: don't deactivate URBs on in-use EP
If an endpoint in use, its associated URBs should not be
deactivated.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 10:55:14 +02:00
Eldad Zack
9372103990 ALSA: usb-audio: remove unused parameter from sync_ep_set_params
Since the format is not actually used in sync_ep_set_params(),
there is no need to pass it down.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 10:52:06 +02:00
Alan Stern
976b6c064a ALSA: improve buffer size computations for USB PCM audio
This patch changes the way URBs are allocated and their sizes are
determined for PCM playback in the snd-usb-audio driver.  Currently
the driver allocates too few URBs for endpoints that don't use
implicit sync, making underruns more likely to occur.  This may be a
holdover from before I/O delays could be measured accurately; in any
case, it is no longer necessary.

The patch allocates as many URBs as possible, subject to four
limitations:

	The total number of URBs for the endpoint is not allowed to
	exceed MAX_URBS (which the patch increases from 8 to 12).

	The total number of packets per URB is not allowed to exceed
	MAX_PACKS (or MAX_PACKS_HS for high-speed devices), which is
	decreased from 20 to 6.

	The total duration of queued data is not allowed to exceed
	MAX_QUEUE, which is decreased from 24 ms to 18 ms.

	The total number of ALSA frames in the output queue is not
	allowed to exceed the ALSA buffer size.

The last requirement is the hardest to implement.  Currently the
number of URBs needed to fill a buffer cannot be determined in
advance, because a buffer contains a fixed number of frames whereas
the number of frames in an URB varies to match shifts in the device's
clock rate.  To solve this problem, the patch changes the logic for
deciding how many packets an URB should contain.  Rather than using as
many as possible without exceeding an ALSA period boundary, now the
driver uses only as many packets as needed to transfer a predetermined
number of frames.  As a result, unless the device's clock has an
exceedingly variable rate, the number of URBs making up each period
(and hence each buffer) will remain constant.

The overall effect of the patch is that playback works better in
low-latency settings.  The user can still specify values for
frames/period and periods/buffer that exceed the capabilities of the
hardware, of course.  But for values that are within those
capabilities, the performance will be improved.  For example, testing
shows that a high-speed device can handle 32 frames/period and 3
periods/buffer at 48 KHz, whereas the current driver starts to get
glitchy at 64 frames/period and 2 periods/buffer.

A side effect of these changes is that the "nrpacks" module parameter
is no longer used.  The patch removes it.

Signed-off-by: Alan Stern <stern@rowland.harvard.edu>
CC: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Daniel Mack <zonque@gmail.com>
Tested-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-26 10:25:31 +02:00
Takashi Iwai
68538bf2bc ASoC: Updates for v3.12
- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
   regressions in the special cases for non-DAPM CODECs and make it
   easier to integrate with other components on boards.  All existing
   drivers have had some level of DAPM support added.
 - A lot of cleanups in DAPM plus support for maintaining controls in a
   specific state while a DAPM widget all contributed by Lars-Peter Clausen.
 - Core helpers for bitbanged AC'97 reset from Markus Pargmann.
 - New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
   Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
   machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
   Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
   Microelectronics WM8997.
 - Support for building drivers that can support it cross-platform for
   compile test.
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Merge tag 'asoc-v3.12' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: Updates for v3.12

- DAPM is now mandatory for CODEC drivers in order to avoid the repeated
  regressions in the special cases for non-DAPM CODECs and make it
  easier to integrate with other components on boards.  All existing
  drivers have had some level of DAPM support added.
- A lot of cleanups in DAPM plus support for maintaining controls in a
  specific state while a DAPM widget all contributed by Lars-Peter Clausen.
- Core helpers for bitbanged AC'97 reset from Markus Pargmann.
- New drivers and support for Analog Devices ADAU1702 and ADAU1401(a),
  Asahi Kasei Microdevices AK4554, Atmel AT91ASM9x5 and WM8904 based
  machines, Freescale S/PDIF and SSI AC'97, Renesas R-Car SoCs, Samsung
  Exynos5420 SoCs, Texas Instruments PCM1681 and PCM1792A and Wolfson
  Microelectronics WM8997.
- Support for building drivers that can support it cross-platform for
  compile test.
2013-08-23 14:12:22 +02:00
Clemens Ladisch
57e6dae108 ALSA: usb-audio: do not trust too-big wMaxPacketSize values
The driver used to assume that the streaming endpoint's wMaxPacketSize
value would be an indication of how much data the endpoint expects or
sends, and compute the number of packets per URB using this value.

However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes,
while only about 88 or 44 bytes are be actually used.  This discrepancy
would result in URBs with far too few packets, which would not work
correctly on the EHCI driver.

To get correct URBs, use wMaxPacketSize only as an upper limit on the
packet size.

Reported-by: James Stone <jamesmstone@gmail.com>
Tested-by: James Stone <jamesmstone@gmail.com>
Cc: <stable@vger.kernel.org> # 2.6.35+
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-08 11:37:34 +02:00
Eldad Zack
e7e58df8ef ALSA: usb-audio: WARN_ON when alts is passed as NULL
Prevent NULL dereference in snd_usb_add_endpoints(), when
alts is passed as NULL. In this case, WARN (since this is
a non-fatal bug) and return NULL ep. Call sites treat a NULL
return value as an error.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:52:27 +02:00
Clemens Ladisch
c75c5ab575 ALSA: USB: adjust for changed 3.8 USB API
The recent changes in the USB API ("implement new semantics for
URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the
default, and changed this flag to mean that URBs can be delayed.
This is not the behaviour wanted by any of the audio drivers because
it leads to discontinuous playback with very small period sizes.
Therefore, our URBs need to be submitted without this flag.

Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org>
Cc: <stable@vger.kernel.org> # 3.8 only
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-29 10:57:35 +02:00
Daniel Mack
d24f5061ee ALSA: snd-usb: add support for DSD DOP stream transport
In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.

The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.

To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.

The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:32 +02:00
Eldad Zack
98ae472b57 ALSA: usb-audio: spelling correction
Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:30:30 +02:00
Eldad Zack
88766f04c4 ALSA: usb-audio: convert list_for_each to entry variant
Change occurances of list_for_each into list_for_each_entry where
applicable.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:30:06 +02:00
Eldad Zack
28acb12014 ALSA: usb-audio: use sender stride for implicit feedback
For implicit feedback endpoints, the number of bytes for each packet
is matched by the corresponding synchronizing endpoint.
The size is calculated by taking the actual size and dividing it by
the stride - currently by the endpoint's stride, but we should use the
synchronization source's stride.
This is evident when the number of channels differ between the
synchronization source and the implicitly fed-back endpoint, as with
M-Audio Fast Track C400 - the synchronization source (capture)
has 4 channels, while the implicit feedback mode endpoint has 6.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:42:54 +01:00
Takashi Iwai
b2eb950de2 ALSA: usb-audio: stop both data and sync endpoints asynchronously
As we are stopping the endpoints asynchronously now, it's better to
trigger the stop of both data and sync endpoints and wait for pending
stopping operations, instead of the sequential trigger-and-wait
procedure.

So the wait argument in snd_usb_endpoint_stop() is dropped, and it's
expected that the caller synchronizes explicitly by calling
snd_usb_endpoint_sync_pending_stop().  (Actually there is only one
place calling this, so it was safe to change.)

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:56 +01:00
Takashi Iwai
ccc1696d52 ALSA: usb-audio: simplify endpoint deactivation code
For further code simplification, drop the conditional call for
usb_kill_urb() with can_wait argument in deactivate_urbs(), and use
only usb_unlink_urb() and wait_clear_urbs() pairs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:54 +01:00
Takashi Iwai
a9bb36261e ALSA: usb-audio: simplify snd_usb_endpoint_start/stop arguments
Reduce the redundant arguments for snd_usb_endpoint_start() and
snd_usb_endpoint_stop().  Also replaced from int to bool.

No functional changes by this commit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:40 +01:00
Takashi Iwai
20d32022a8 ALSA: usb-audio: Deprecate async_unlink option
The async unlink behavior has been working over years.  The option was
provided only as a workaround for 2.4.x kernel.  Let's get rid of it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:37:40 +01:00
Joe Perches
190006f9d6 ALSA: usb-audio: use bitmap_weight
Use bitmap_weight to count the total number of bits set in bitmap.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-17 11:35:07 +01:00
Takashi Iwai
f58161ba1b ALSA: usb-audio: Fix crash at re-preparing the PCM stream
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback.  It turned out that the problem is that we don't
wait until all URBs are killed.

This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181

Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-08 08:56:44 +01:00
Linus Torvalds
f5a246eab9 Sound updates for 3.7-rc1
This contains pretty many small commits covering fairly large range of
 files in sound/ directory.  Partly because of additional API support
 and partly because of constantly developed ASoC and ARM stuff.
 
 Some highlights:
 
 - Introduced the helper function and documentation for exposing the
   channel map via control API, as discussed in Plumbers; most of PCI
   drivers are covered, will follow more drivers later
 
 - Most of drivers have been replaced with the new PM callbacks (if
   the bus is supported)
 
 - HD-audio controller got the support of runtime PM and the support of
   D3 clock-stop.  Also changing the power_save option in sysfs kicks
   off immediately to enable / disable the power-save mode.
 
 - Another significant code change in HD-audio is the rewrite of
   firmware loading code.  Other than that, most of changes in HD-audio
   are continued cleanups and standardization for the generic auto
   parser and bug fixes (HBR, device-specific fixups), in addition to
   the support of channel-map API.
 
 - Addition of ASoC bindings for the compressed API, used by the
   mid-x86 drivers.
 
 - Lots of cleanups and API refreshes for ASoC codec drivers and
   DaVinci.
 
 - Conversion of OMAP to dmaengine.
 
 - New machine driver for Wolfson Microelectronics Bells.
 
 - New CODEC driver for Wolfson Microelectronics WM0010.
 
 - Enhancements to the ux500 and wm2000 drivers
 
 - A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...
2012-10-09 07:07:14 +09:00
Daniel Mack
8dce30c891 ALSA: snd-usb: fix next_packet_size calls for pause case
Also fix the calls to next_packet_size() for the pause case. This was
missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size").

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de>
Cc: stable@kernel.org
[ Taking directly because Takashi is on vacation  - Linus ]
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2012-09-27 16:46:15 -07:00
Dylan Reid
35ec7aa298 ALSA: usb-audio: Don't require hw_params in endpoint.
Change the interface to configure an endpoint so that it doesn't require
a hw_params struct.  This will allow it to be called from prepare
instead of hw_params, configuring it after system resume.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:07:52 +02:00
Takashi Iwai
0528842690 Merge branch 'for-linus' into for-next
To merge HD-audio fixes back to 3.7 development line
2012-09-11 16:46:36 +02:00
Daniel Mack
2b58fd5b31 ALSA: snd-usb: Add quirks for Playback Designs devices
Playback Designs' USB devices have some hardware limitations on their
USB interface. In particular:

 - They need a 20ms delay after each class compliant request as the
   hardware ACKs the USB packets before the device is actually ready
   for the next command. Sending data immediately will result in buffer
   overflows in the hardware.
 - The devices send bogus feedback data at the start of each stream
   which confuse the feedback format auto-detection.

This patch introduces a new quirks hook that is called after each
control packet and which adds a delay for all devices that match
Playback Designs' USB VID for now.

In addition, it adds a counter to snd_usb_endpoint to drop received
packets on the floor. Another new quirks function that is called once
an endpoint is started initializes that counter for these devices on
their sync endpoint.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com>
Supported-by: Demian Martin <demianm_1@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-04 11:31:14 +02:00
Daniel Mack
245baf983c ALSA: snd-usb: fix calls to next_packet_size
In order to support devices with implicit feedback streaming models,
packet sizes are now stored with each individual urb, and the PCM
handling code which fills the buffers purely relies on the size fields
now.

However, calling snd_usb_audio_next_packet_size() for all possible
packets in an URB at once, prior to letting the PCM code do its job
does in fact not lead to the same behaviour than what the old code did:
The PCM code will break its loop once a period boundary is reached,
consequently using up less packets that it really could.

As snd_usb_audio_next_packet_size() implements a feedback mechanism to
the endpoints phase accumulator, the number of calls to that function
matters, and when called too often, the data rate runs out of bounds.

Fix this by making the next_packet function public, and call it from the
PCM code as before if the packet data sizes are not defined.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:03:48 +02:00
Daniel Mack
015618b902 ALSA: snd-usb: Fix URB cancellation at stream start
Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in
PCM capture stream") fixed a scheduling-while-atomic bug that happened
when snd_usb_endpoint_start was called from the trigger callback, which
is an atmic context. However, the patch breaks the idea of the endpoints
reference counting, which is the reason why the driver has been
refactored lately.

Revert that commit and let snd_usb_endpoint_start() take care of the URB
cancellation again. As this function is called from both atomic and
non-atomic context, add a flag to denote whether the function may sleep.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-30 07:46:27 +02:00