1. 19 10月, 2015 1 次提交
  2. 13 10月, 2015 1 次提交
    • R
      ALSA: usb-audio: Fix max packet size calculation for USB audio · ab30965d
      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
  3. 26 8月, 2015 1 次提交
    • T
      ALSA: usb-audio: Avoid nested autoresume calls · 47ab1545
      Takashi Iwai 提交于
      After the recent fix of runtime PM for USB-audio driver, we got a
      lockdep warning like:
      
        =============================================
        [ INFO: possible recursive locking detected ]
        4.2.0-rc8+ #61 Not tainted
        ---------------------------------------------
        pulseaudio/980 is trying to acquire lock:
         (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
        but task is already holding lock:
         (&chip->shutdown_rwsem){.+.+.+}, at: [<ffffffffa0355dac>] snd_usb_autoresume+0x1d/0x52 [snd_usb_audio]
      
      This comes from snd_usb_autoresume() invoking down_read() and it's
      used in a nested way.  Although it's basically safe, per se (as these
      are read locks), it's better to reduce such spurious warnings.
      
      The read lock is needed to guarantee the execution of "shutdown"
      (cleanup at disconnection) task after all concurrent tasks are
      finished.  This can be implemented in another better way.
      
      Also, the current check of chip->in_pm isn't good enough for
      protecting the racy execution of multiple auto-resumes.
      
      This patch rewrites the logic of snd_usb_autoresume() & co; namely,
      - The recursive call of autopm is avoided by the new refcount,
        chip->active.  The chip->in_pm flag is removed accordingly.
      - Instead of rwsem, another refcount, chip->usage_count, is introduced
        for tracking the period to delay the shutdown procedure.  At
        the last clear of this refcount, wake_up() to the shutdown waiter is
        called.
      - The shutdown flag is replaced with shutdown atomic count; this is
        for reducing the lock.
      - Two new helpers are introduced to simplify the management of these
        refcounts; snd_usb_lock_shutdown() increases the usage_count, checks
        the shutdown state, and does autoresume.  snd_usb_unlock_shutdown()
        does the opposite.  Most of mixer and other codes just need this,
        and simply returns an error if it receives an error from lock.
      
      Fixes: 9003ebb1 ('ALSA: usb-audio: Fix runtime PM unbalance')
      Reported-and-tested-by: NAlexnader Kuleshov <kuleshovmail@gmail.com>
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      47ab1545
  4. 10 11月, 2014 1 次提交
    • T
      ALSA: pcm: Add snd_pcm_stop_xrun() helper · 1fb8510c
      Takashi Iwai 提交于
      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: NTakashi Iwai <tiwai@suse.de>
      1fb8510c
  5. 06 11月, 2014 1 次提交
    • T
      ALSA: usb-audio: Trigger PCM XRUN at XRUN · 67e22500
      Takashi Iwai 提交于
      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: NTakashi Iwai <tiwai@suse.de>
      67e22500
  6. 04 11月, 2014 1 次提交
  7. 26 6月, 2014 1 次提交
    • T
      ALSA: usb-audio: Fix races at disconnection and PCM closing · 92a586bd
      Takashi Iwai 提交于
      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: NAlan Stern <stern@rowland.harvard.edu>
      Cc: <stable@vger.kernel.org>
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      92a586bd
  8. 03 5月, 2014 1 次提交
  9. 26 2月, 2014 1 次提交
    • T
      ALSA: usb-audio: Use standard printk helpers · 0ba41d91
      Takashi Iwai 提交于
      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: NTakashi Iwai <tiwai@suse.de>
      0ba41d91
  10. 27 11月, 2013 1 次提交
  11. 07 10月, 2013 5 次提交
  12. 26 9月, 2013 1 次提交
    • A
      ALSA: improve buffer size computations for USB PCM audio · 976b6c06
      Alan Stern 提交于
      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: NAlan Stern <stern@rowland.harvard.edu>
      CC: Clemens Ladisch <clemens@ladisch.de>
      Tested-by: NDaniel Mack <zonque@gmail.com>
      Tested-by: NEldad Zack <eldad@fogrefinery.com>
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      976b6c06
  13. 08 8月, 2013 1 次提交
  14. 06 8月, 2013 1 次提交
  15. 29 4月, 2013 1 次提交
  16. 18 4月, 2013 1 次提交
    • D
      ALSA: snd-usb: add support for DSD DOP stream transport · d24f5061
      Daniel Mack 提交于
      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: NDaniel Mack <zonque@gmail.com>
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      d24f5061
  17. 04 4月, 2013 2 次提交
  18. 29 11月, 2012 1 次提交
    • E
      ALSA: usb-audio: use sender stride for implicit feedback · 28acb120
      Eldad Zack 提交于
      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: NEldad Zack <eldad@fogrefinery.com>
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      28acb120
  19. 21 11月, 2012 4 次提交
  20. 17 11月, 2012 1 次提交
  21. 08 11月, 2012 1 次提交
  22. 28 9月, 2012 1 次提交
  23. 19 9月, 2012 1 次提交
  24. 04 9月, 2012 1 次提交
    • D
      ALSA: snd-usb: Add quirks for Playback Designs devices · 2b58fd5b
      Daniel Mack 提交于
      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: NDaniel Mack <zonque@gmail.com>
      Reported-and-tested-by: NAndreas Koch <andreas@akdesigninc.com>
      Supported-by: NDemian Martin <demianm_1@yahoo.com>
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      2b58fd5b
  25. 01 9月, 2012 1 次提交
    • D
      ALSA: snd-usb: fix calls to next_packet_size · 245baf98
      Daniel Mack 提交于
      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: NDaniel Mack <zonque@gmail.com>
      Cc: stable@kernel.org [v3.5+]
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      245baf98
  26. 30 8月, 2012 1 次提交
    • D
      ALSA: snd-usb: Fix URB cancellation at stream start · 015618b9
      Daniel Mack 提交于
      Commit e9ba389c ("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: NDaniel Mack <zonque@gmail.com>
      Cc: stable@kernel.org [3.5+]
      Signed-off-by: NTakashi Iwai <tiwai@suse.de>
      015618b9
  27. 16 8月, 2012 1 次提交
  28. 13 7月, 2012 1 次提交
  29. 25 4月, 2012 1 次提交
  30. 24 4月, 2012 1 次提交
  31. 13 4月, 2012 2 次提交