Commit Graph

95 Commits

Author SHA1 Message Date
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 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
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
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
Pierre-Louis Bossart ea33d359c4 ALSA: usb: update trigger timestamp on first non-zero URB submitted
The first URBs are submitted during the prepare stage. When .trigger is
called, the ALSA core saves a trigger tstamp that doesn't correspond to
the actual time when the samples are submitted. The trigger_tstamp is
now updated when the first data are submitted to avoid any time offsets.

A usb-specific trigger_tstamp_pending_update flag is used for now,
at some point the flag would need to move to the ALSA core, USB
is not the only interface where silent block transfers are programmed
as part of the prepare stage, with actual data enabled when .trigger
is called.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-02-09 16:02:43 +01:00
Jurgen Kramer 6874daad4b ALSA: usb-audio: Add mode select quirk for Denon/Marantz DACs
Denon/Marantz USB DACs need a specific vendor command to switch between PCM and
DSD mode. This patch adds a new quirk function to switch between the two modes
using the specific USB vendor command.

This patch applies to the following devices:
- Marantz SA-14S1
- Marantz HD-DAC1

Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-28 18:02:35 +01:00
Sander Eikelenboom b7a7723513 ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while DEBUG not defined
This (widely used) construction:

if(printk_ratelimit())
	dev_dbg()

Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.

[  533.803964] retire_playback_urb: 852 callbacks suppressed
[  538.807930] retire_playback_urb: 852 callbacks suppressed
[  543.811897] retire_playback_urb: 852 callbacks suppressed
[  548.815745] retire_playback_urb: 852 callbacks suppressed
[  553.819826] retire_playback_urb: 852 callbacks suppressed

So use dev_dbg_ratelimited() instead of this construction.

Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02 18:10:59 +02:00
Tim Gardner a5065eb6da ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()
BugLink: http://bugs.launchpad.net/bugs/1305133

Malfunctioning or slow devices can cause a flood of dmesg SPAM.

I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour
of prior art in sound/usb/pcm.c.

WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit
+	if (printk_ratelimit() &&

Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Eldad Zack <eldad@fogrefinery.com>
Cc: Daniel Mack <zonque@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-09 21:07:38 +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
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 06613f547a ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error
If setting the interface fails, the SUBSTREAM_FLAG_SYNC_EP_STARTED
should be cleared.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 11:00:23 +02:00
Eldad Zack 26de5d0a8d ALSA: usb-audio: remove deactivate_endpoints()
The only call site for deactivate_endpoints() at snd_usb_hw_free().
The return value is not checked there, as it is irrelevant if it
fails on hw_free.
This patch moves the deactivation of the endpoints directly into
snd_usb_hw_free().

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 10:52:13 +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
Eldad Zack 88abb8eff4 ALSA: usb-audio: remove implicit_fb from quirk
Since the quirks all apply to implicit feedback (the source endpoint
is always a data endpoint), there's no need to set and check
a flag for it.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:52:14 +02:00
Eldad Zack 914273c714 ALSA: usb-audio: remove is_playback from implicit feedback quirks
An implicit feedback endpoint can only be a capture source. The
consumer (sink) of the implicit feedback endpoint is therefore limited
to playback EPs.
Check if the target endpoint is a playback first and remove redundant
checks.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:51:48 +02:00
Eldad Zack 95fec88332 ALSA: usb-audio: do not initialize and check implicit_fb
Since implicit_fb is not changed, !implicit_fb will always
be true - it is set only after these checks.
Similarly, there's also no need to set it at the top of the function.

Change the type of implicit_fb to bool (more appropriate).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:51:11 +02:00
Eldad Zack f34d065013 ALSA: usb-audio: reverse condition logic in set_sync_endpoint
Reverse logic on the conditions required to qualify for a sync endpoint
and remove one level of indendation.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:50:15 +02:00
Eldad Zack a60945fd08 ALSA: usb-audio: move implicit fb quirks to separate function
Separate setting implicit feedback quirks from setting
a sync endpoint (which may also be explicit feedback or async).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:49:21 +02:00
Eldad Zack 71bb64c56d ALSA: usb-audio: separate sync endpoint setting from set_format
Setting the sync endpoint currently takes up about half of set_format().
Move it to a dedicated function.
No functional change.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:48:34 +02:00
Eldad Zack d133f2c22e ALSA: usb-audio: remove assignment from if condition
Following general kernel style.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:48:22 +02:00
Eldad Zack d833cdb10c ALSA: usb-audio: remove disabled debug code in set_format
Code block does not compile when enabled.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-06 10:48:12 +02:00
Clemens Ladisch ba7c2be114 ALSA: usb-audio: detect implicit feedback on Roland devices
All the Roland/Edirol/BOSS USB audio devices that need implicit feedback
show this unambiguously in their descriptors, so it might be a good idea
to let the driver detect this.

This should make playback work correctly (at least with Jack) with the
following devices:
- BOSS GT-100
- BOSS JS-8 Jam Station
- Edirol M-16DX
- Roland GAIA SH-01

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:47 +02:00
Clemens Ladisch 8f898e92ae ALSA: usb-audio: store protocol version in struct audioformat
Instead of reading bInterfaceProtocol from the descriptor whenever it's
needed, store this value in the audioformat structure.  Besides
simplifying some code, this will allow us to correctly handle vendor-
specific devices where the descriptors are marked with other values.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:47 +02:00
Eldad Zack 74c34ca1cc ALSA: pcm_format_to_bits strong-typed conversion
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.

Change such conversions to use this function and silence sparse
warnings.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-29 13:36:15 +02:00
Daniel Mack 44dcbbb1cd ALSA: snd-usb: add support for bit-reversed byte formats
There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.

ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.

This patch adds support for this by adding a boolean flag to the
audio format struct.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:47 +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
Daniel Mack 8a2a74d2b7 ALSA: snd-usb: use ep->stride from urb callbacks
For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:23 +02:00
Calvin Owens 1539d4f82a ALSA: usb: Add quirk for 192KHz recording on E-Mu devices
When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.

Userspace expected:  L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1

Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.

Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.

Signed-off-by: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-13 10:58:03 +02:00
Daniel Mack 21bb5aafce ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locations
It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-10 09:21:43 +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
Daniel Mack 0959f22ee6 ALSA: snd-usb: add delay quirk for "Playback Design" products
"Playback Design" products need a 50ms delay after setting the USB
interface.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:47:21 +01:00
Matt Gruskin e9a25e04b8 ALSA: usb-audio: add support for M-Audio FT C600
Adds quirks and mixer support for the M-Audio Fast Track C600 USB
audio interface. This device is very similar to the C400 - the C600
simply has some more inputs and outputs, so the existing C400 support
is extended to support this device as well.

Signed-off-by: Matt Gruskin <matthew.gruskin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-02-11 14:02:27 +01:00
Takashi Iwai 86b2723725 ALSA: Make snd_printd() and snd_printdd() inline
Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
  sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
  sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]

For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros.  This should have the same effect but shut up warnings like
above.

But since we had already put ifdefs, changing to inline functions
would trigger compile errors.  So, such ifdefs is removed in this
patch.

In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too.  For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.

Reported-by: Stratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-25 18:32:14 +01:00
Takashi Iwai e152f18027 Merge branch 'for-linus' into for-next
This is a preliminary merge before the upcoming merge of generic parser
branch.
2013-01-23 08:31:34 +01:00
Takashi Iwai 31be5425d7 ALSA: usb-audio: Fix NULL dereference by access to non-existing substream
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP.  But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.

This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.

Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-11 11:12:17 +01:00
Pierre-Louis Bossart e4cc615340 ALSA: usb-audio: support delay calculation on capture streams
Enable delay report on capture path. The delay is reset when an
URB is retired and increment at each call to .pointer based
on frame counter changes. The precision of the delay
information is limited to 1ms as in the playback case.

This reverts commit 3f94fad095.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-24 10:53:57 +01:00
Eldad Zack 0d9741c0e0 ALSA: usb-audio: sync ep init fix for audioformat mismatch
Commit 947d299686 , "ALSA: snd-usb:
properly initialize the sync endpoint", while correcting the
initialization of the sync endpoint when opening just the data
endpoint, prevents devices that has a sync endpoint, with a channel
number different than that of the data endpoint, from functioning.
Due to a different channel and period bytes count, attempting to
initialize the sync endpoint will fail at the usb host driver.
For example, when using xhci:

 cannot submit urb 0, error -90: internal error

With this patch, if a sync endpoint has multiple audioformats, a
matching audioformat is preferred. An audioformat must be found
with at least one channel and support the requested sample rate
and PCM format, otherwise the stream will not be opened.

If the number of channels differ between the selected audioformat
and the requested format, adjust the period bytes count accordingly.
It is safe to perform the calculation on the basis of the channel
count, since the requested PCM audio format and the rate must be
supported by the selected audioformat.

Cc: Jeffrey Barish <jeff_barish@earthlink.net>
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04 08:14:31 +01:00
Eldad Zack ca10a7ebdf ALSA: usb-audio: FT C400 sync playback EP to capture EP
The playback endpoint uses implicit feedback mode, similar
to the M-Audio FTU. Like with the FTU, we need to associate
the sync pipe ourselves.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:45:18 +01:00
Eldad Zack fde854bdaf ALSA: usb-audio: replace hardcoded value with const
In this context, 0x01 is USB_ENDPOINT_XFER_ISOC.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:42:33 +01:00
Takashi Iwai 48779a0b8f ALSA: usb-audio: fix delay account during pause
When a playback stream is paused, the stream isn't actually stopped,
thus we still need to take care of the in-flight data amount for the
delay calculation.  Otherwise the value of subs->last_delay is no
longer reliable and can give a bogus value after resuming from pause.
This will result in "delay: estimated XX, actual YY" error messages.

Also, during pause after all in flight data are processed
(i.e. last_delay = 0), we don't have to calculate the actual delay
from the current frame.  Give a short path in such a case.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23 16:07:11 +01:00
Takashi Iwai 3f94fad095 ALSA: usb-audio: ignore delay calculation for capture stream
It doesn't make sense to calculate the delay for capture streams in
the current implementation.  It's always zero, so we should skip the
computation in snd_usb_pcm_pointer() in the case of capture.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-23 15:37:32 +01:00
Takashi Iwai 2ba509a6ba Merge branch 'for-linus' into for-next 2012-11-22 21:22:39 +01:00
Daniel Mack 947d299686 ALSA: snd-usb: properly initialize the sync endpoint
Jeffrey Barish reported an obvious bug in the pcm part of the usb-audio
driver which causes the code to not initialize the sync endpoint from
configure_endpoint().

Reported-by: Jeffrey Barish <jeff_barish@earthlink.net>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-22 21:22:33 +01:00
Takashi Iwai b0db6063db ALSA: usb-audio: process pending stop at PCM hw_free and close
PCM hw_free and close should wait until all the pending stop
operations have been finished.  Basically only PCM trigger callback
should use non-wait calls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:43:58 +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 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