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  1. 15 10月, 2015 3 次提交
  3. 13 10月, 2015 4 次提交
    • R
      ALSA: usb-audio: Fix max packet size calculation for USB audio · ab30965d
      由 Ricard Wanderlof 提交于 
      Rounding must take place before multiplication with the frame size, since
each packet contains a whole number of frames.

We must also properly consider the data interval, as a larger data
interval will result in larger packets, which, depending on the sampling
frequency, can result in packet sizes that are less than integral
multiples of the packet size for a lower data interval.

Detailed explanation and rationale:

The code before this commit had the following expression on line 613 to
calculate the maximum isochronous packet size:

	maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3))
			>> (16 - ep->datainterval);

Here, ep->freqmax is the maximum assumed sample frequency, calculated from the
nominal sample frequency plus 25%. It is ultimately derived from ep->freqn,
which is in the units of frames per packet, from get_usb_full_speed_rate()
or usb_high_speed_rate(), as applicable, in Q16.16 format.

The expression essentially adds the Q16.16 equivalent of 0.999... (i.e.
the largest number less than one) to the sample rate, in order to get a
rate whose integer part is rounded up from the fractional value. The
multiplication with (frame_bits >> 3) yields the number of bytes in a
packet, and the (16 >> ep->datainterval) then converts it from Q16.16 back
to an integer, taking into consideration the bDataInterval field of the
endpoint descriptor (which describes how often isochronous packets are
transmitted relative to the (micro)frame rate (125us or 1ms, for USB high
speed and full speed, respectively)). For this discussion we will initially
assume a bDataInterval of 0, so the second line of the expression just
converts the Q16.16 value to an integer.

In order to illustrate the problem, we will set frame_bits 64, which
corresponds to a frame size of 8 bytes.

The problem here is twofold. First, the rounding operation consists
of the addition of 0x0.ffff and subsequent conversion to integer, but as the
expression stands, the conversion to integer is done after multiplication
with the frame size, rather than before. This results in the resulting
maxsize becoming too large.

Let's take an example. We have a sample rate of 96 kHz, so our ep->freqn is
0xc0000 (see usb_high_speed_rate()). Add 25% (line 612) and we get 0xf0000.
The calculated maxsize is then ((0xf0000 + 0x0ffff) * 8) >> 16 = 127 .
However, if we do the number of bytes calculation in a less obscure way it's
more apparent what the true corresponding packet size is: we get
ceil(96000 * 1.25 / 8000) * 8 = 120, where 1.25 is the 25% from line 612,
and the 8000 is the number of isochronous packets per second on a high
speed USB connection (125 us microframe interval).

This is fixed by performing the complete rounding operation prior to
multiplication with the frame rate.

The second problem is that when considering the ep->datainterval, this
must be done before rounding, in order to take the advantage of the fact
that if the number of bytes per packet is not an integer, the resulting
rounded-up integer is not necessarily a factor of two when the data
interval is increased by the same factor.

For instance, assuming a freqency of 41 kHz, the resulting
bytes-per-packet value for USB high speed is 41 kHz / 8000 = 5.125, or
0x52000 in Q16.16 format. With a data interval of 1 (ep->datainterval = 0),
this means that 6 frames per packet are needed, whereas with a data
interval of 2 we need 10.25, i.e. 11 frames needed.

Rephrasing the maxsize expression to:

	maxsize = (((ep->freqmax << ep->datainterval) + 0xffff) >> 16) *
			 (frame_bits >> 3);

for the above 96 kHz example we instead get
((0xf0000 + 0xffff) >> 16) * 8 = 120 which is the correct value.

We can also do the calculation with a non-integer sample rate which is when
rounding comes into effect: say we have 44.1 kHz (resulting ep->freqn =
0x58333, and resulting ep->freqmax 0x58333 * 1.25 = 0x6e3ff (rounded down)):

Original maxsize = ((0x6e3ff + 0xffff) * 8) << 16 = 63 (63.124.. rounded down)
True maxsize = ceil(44100 * 1.25 / 8000) * 8 = 7 * 8 = 56
New maxsize = ((0x6e3ff + 0xffff) >> 16) * 8 = 7 * 8 = 56

This is also corroborated by the wMaxPacketSize check on line 616. Assume
that wMaxPacketSize = 104, with ep->maxpacksize then having the same value.
As 104 < 127, we get maxsize = 104. ep->freqmax is then recalculated to
(104 / 8) << 16 = 0xd0000 . Putting that rate into the original maxsize
calculation yields a maxsize of ((0xd0000 + 0xffff) * 8) >> 16 = 111
(with decimals 111.99988). Clearly, we should get back the 104 here,
which we would with the new expression: ((0xd0000 + 0xffff) >> 16) * 8 = 104 .

(The error has not been a problem because it only results in maxsize being
a bit too big which just wastes a couple of bytes, either as a result of
the first maxsize calculation, or because the resulting calculation will
hit the wMaxPacketSize value before the packet is too big, resulting in
fixing the size to wMaxPacketSize even though the packet is actually not
too long.)

Tested with an Edirol UA-5 both at 44.1 kHz and 96 kHz.
Signed-off-by: NRicard Wanderlof <ricardw@axis.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      ab30965d
    • T
      Merge branch 'for-linus' into for-next · 3c69ea44
      由 Takashi Iwai 提交于
      3c69ea44
    • D
      ALSA: hda - Fix inverted internal mic on Lenovo G50-80 · e8d65a8d
      由 David Henningsson 提交于 
      Add the appropriate quirk to indicate the Lenovo G50-80 has a stereo
mic input where one channel has reverse polarity.

Alsa-info available at:
https://launchpadlibrarian.net/220846272/AlsaInfo.txt

Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1504778Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      e8d65a8d
    • V
      ALSA: hdac: Explicitly add io.h · 42f2bb1c
      由 Vinod Koul 提交于 
      Compiling the hdac extended core on arm fails with below error:

  sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_writel':
>> sound/hda/ext/hdac_ext_bus.c:29:2: error: implicit declaration of
>> function
+'writel' [-Werror=implicit-function-declaration]
     writel(value, addr);
     ^
   sound/hda/ext/hdac_ext_bus.c: In function 'hdac_ext_readl':
>> sound/hda/ext/hdac_ext_bus.c:34:2: error: implicit declaration of
>> function
+'readl' [-Werror=implicit-function-declaration]
     return readl(addr);

This is fixed by explicitly including io.h

Fixes: 99463b3a - ('ALSA: hda: provide default bus io ops extended hdac')
Reported-by: Nkbuild test robot <lkp@intel.com>
Suggested-by: NMark Brown <broonie@kernel.org>
Signed-off-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      42f2bb1c
  5. 12 10月, 2015 14 次提交
  7. 09 10月, 2015 8 次提交
    • T
      ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error · bde3e288
      由 Takashi Sakamoto 提交于 
      Currently, when asynchronous transactions finish in error state and
retries, work scheduling and work running also continues. This
should be canceled at fatal error because it can cause endless loop.

This commit enables to cancel transferring MIDI messages when transactions
encounter fatal errors. This is achieved by setting error state.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      bde3e288
    • T
      ALSA: firewire-lib: add throttle for MIDI data rate · ea848b7b
      由 Takashi Sakamoto 提交于 
      Typically, the target devices have internal buffer to adjust output of
received MIDI messages for MIDI serial bus, while the capacity of the
buffer is limited. IEEE 1394 transactions can transfer more MIDI messages
than MIDI serial bus can. This can cause buffer over flow in device side.

This commit adds throttle to limit MIDI data rate by counting intervals
between two MIDI messages. Usual MIDI messages consists of two or three
bytes. This requires 1.302 to 1.953 mili-seconds interval between these
messages. This commit uses kernel monotonic time service to calculate the
time of next transaction.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      ea848b7b
    • T
      ALSA: firewire-lib: schedule work again when MIDI substream has rest of MIDI messages · e8a40d9b
      由 Takashi Sakamoto 提交于 
      Currently, when two MIDI trigger callbacks can be called immediately,
transactions for the second MIDI messages can be postpone till next trigger
callback. This is not good for real-time message transmission.

This commit schedules work again at response handling callback if the
MIDI substream still includes untransferred MIDI messages.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      e8a40d9b
    • T
      ALSA: firewire-lib: add a restriction for a transaction at once · d3ef9cb9
      由 Takashi Sakamoto 提交于 
      Currently, when waiting for a response, callers can start another
transaction by scheduling another work. This is not good for error
processing of transaction, especially the first response is too late.

This commit serialize request/response transactions, by adding one
boolean member to represent idling state.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      d3ef9cb9
    • T
      ALSA: firewire-lib: add helper functions for asynchronous transactions to transfer MIDI messages · 585d7cba
      由 Takashi Sakamoto 提交于 
      Some models receive MIDI messages via IEEE 1394 asynchronous transactions.
In this case, MIDI messages are transferred in fixed-length payload. It's
nice that firewire-lib module has common helper functions.

This commit implements this idea. Each driver adds
'struct snd_fw_async_midi_port' in its instance structure. In probing,
it should call snd_fw_async_midi_port_init() to initialize the
structure with some parameters such as target address, the length
of payload in a transaction and a pointer for callback function
to fill the payload buffer. At 'struct snd_rawmidi_ops.trigger()'
callback, it should call 'snd_fw_async_midi_port_run()' to start
transactions. Each driver should ensure that the lifetime of MIDI
substream continues till calling 'snd_fw_async_midi_port_finish()'.

The helper functions support retries to transferring MIDI messages when
transmission errors occur. When transactions are successful, the helper
functions call 'snd_rawmidi_transmit_ack()' internally to consume MIDI
bytes in the buffer. Therefore, Each driver is expected to use
'snd_rawmidi_transmit_peek()' to tell the number of bytes to transfer to
return value of 'fill' callback.
Signed-off-by: NTakashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      585d7cba
    • K
      ALSA: seq_oss: fix waitqueue_active without memory barrier in snd-seq-oss · 69447027
      由 Kosuke Tatsukawa 提交于 
      snd_seq_oss_readq_put_event() seems to be missing a memory barrier which
might cause the waker to not notice the waiter and miss sending a
wake_up as in the following figure.

    snd_seq_oss_readq_put_event		    snd_seq_oss_readq_wait
------------------------------------------------------------------------
					/* wait_event_interruptible_timeout */
					 /* __wait_event_interruptible_timeout */
					  /* ___wait_event */
					  for (;;) {									 prepare_to_wait_event(&wq, &__wait,
					    state);
spin_lock_irqsave(&q->lock, flags);
if (waitqueue_active(&q->midi_sleep))
/* The CPU might reorder the test for
   the waitqueue up here, before
   prior writes complete */
					  if ((q->qlen>0 || q->head==q->tail)
					  ...
					  __ret = schedule_timeout(__ret)
if (q->qlen >= q->maxlen - 1) {
memcpy(&q->q[q->tail], ev, sizeof(*ev));
q->tail = (q->tail + 1) % q->maxlen;
q->qlen++;
------------------------------------------------------------------------

There are two other place in sound/core/seq/oss/ which have similar
code.  The attached patch removes the call to waitqueue_active() leaving
just wake_up() behind.  This fixes the problem because the call to
spin_lock_irqsave() in wake_up() will be an ACQUIRE operation.

I found this issue when I was looking through the linux source code
for places calling waitqueue_active() before wake_up*(), but without
preceding memory barriers, after sending a patch to fix a similar
issue in drivers/tty/n_tty.c  (Details about the original issue can be
found here: https://lkml.org/lkml/2015/9/28/849).
Signed-off-by: NKosuke Tatsukawa <tatsu@ab.jp.nec.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      69447027
    • V
      ALSA: hda: make use of core codec fns · 70b4891c
      由 Vinod Koul 提交于 
      Now that we have introduced the core fns we should make hda use these
helpers
Signed-off-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      70b4891c
    • S
      ALSA: hdac: Copy codec helpers to core · 1b5e6167
      由 Subhransu S. Prusty 提交于 
      The current codec helpers are local to hda code and needs to be moved to
core so that other users can use it.
The helpers to read/write the codec and to check the
power state of widgets is copied
Signed-off-by: NSubhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      1b5e6167
  9. 08 10月, 2015 1 次提交
  11. 07 10月, 2015 2 次提交
  13. 06 10月, 2015 2 次提交
  15. 05 10月, 2015 4 次提交
  17. 04 10月, 2015 2 次提交
    • L
      Linux 4.3-rc4 · 049e6dde
      由 Linus Torvalds 提交于
      049e6dde
    • L
      Merge branch 'strscpy' of git://git.kernel.org/pub/scm/linux/kernel/git/cmetcalf/linux-tile · 30c44659
      由 Linus Torvalds 提交于 
      Pull strscpy string copy function implementation from Chris Metcalf.

Chris sent this during the merge window, but I waffled back and forth on
the pull request, which is why it's going in only now.

The new "strscpy()" function is definitely easier to use and more secure
than either strncpy() or strlcpy(), both of which are horrible nasty
interfaces that have serious and irredeemable problems.

strncpy() has a useless return value, and doesn't NUL-terminate an
overlong result.  To make matters worse, it pads a short result with
zeroes, which is a performance disaster if you have big buffers.

strlcpy(), by contrast, is a mis-designed "fix" for strlcpy(), lacking
the insane NUL padding, but having a differently broken return value
which returns the original length of the source string.  Which means
that it will read characters past the count from the source buffer, and
you have to trust the source to be properly terminated.  It also makes
error handling fragile, since the test for overflow is unnecessarily
subtle.

strscpy() avoids both these problems, guaranteeing the NUL termination
(but not excessive padding) if the destination size wasn't zero, and
making the overflow condition very obvious by returning -E2BIG.  It also
doesn't read past the size of the source, and can thus be used for
untrusted source data too.

So why did I waffle about this for so long?

Every time we introduce a new-and-improved interface, people start doing
these interminable series of trivial conversion patches.

And every time that happens, somebody does some silly mistake, and the
conversion patch to the improved interface actually makes things worse.
Because the patch is mindnumbing and trivial, nobody has the attention
span to look at it carefully, and it's usually done over large swatches
of source code which means that not every conversion gets tested.

So I'm pulling the strscpy() support because it *is* a better interface.
But I will refuse to pull mindless conversion patches.  Use this in
places where it makes sense, but don't do trivial patches to fix things
that aren't actually known to be broken.

* 'strscpy' of git://git.kernel.org/pub/scm/linux/kernel/git/cmetcalf/linux-tile:
  tile: use global strscpy() rather than private copy
  string: provide strscpy()
  Make asm/word-at-a-time.h available on all architectures
      30c44659