Commit Graph

1144 Commits

Author SHA1 Message Date
Sachin Kamat 6b5a7c66ce ALSA: usb-audio: Use module_usb_driver
module_usb_driver makes code simpler by removing the boilerplate.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-09 14:00:34 +02:00
Takashi Iwai d820306cbe Merge branch 'for-linus' into for-next
For updating the HDMI chmap fix.

Conflicts:
	sound/pci/hda/patch_hdmi.c
2013-10-08 09:30:04 +02:00
Thomas Pugliese 6d5eba5aac ALSA: usb-audio: support wireless devices in snd_usb_parse_datainterval
This patch adds support for dev speed USB_SPEED_WIRELESS in
snd_usb_parse_datainterval which allows the usb sound core to create
ISO urbs with the correct number and size of buffers.

Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 12:52:21 +02:00
Thomas Pugliese df3774c5c5 ALSA: usb-audio: add support for wireless USB devices
This patch updates snd_usb_audio_create also support devices whose
speed == USB_SPEED_WIRELESS.

Signed-off-by: Thomas Pugliese <thomas.pugliese@gmail.com>
Acked-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-07 12:51:54 +02:00
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 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 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 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
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
Daniel Mack a9d14bc0b1 ALSA: snd-usb-usx2y: remove bogus frame checks
The frame check in i_usX2Y_urb_complete() and
i_usX2Y_usbpcm_urb_complete() is bogus and produces false positives as
described in this LAU thread:

  http://linuxaudio.org/mailarchive/lau/2013/5/20/200177

This patch removes the check code entirely.

Cc: fzu@wemgehoertderstaat.de
Reported-by: Dr Nicholas J Bailey <nicholas.bailey@glasgow.ac.uk>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-10-02 17:58:01 +02:00
Hannes Gräuler d2724de187 ALSA: snd-usb-caiaq: LED support for Maschine Controller
This patch adds LED support for the Native Instruments Maschine
Controller. It adds ALSA controls for dimming the LEDs of all
buttons and the backlight of the two displays.

Signed-off-by: Hannes Gräuler <hgraeule@uos.de>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-30 11:19:16 +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
Peter Senna Tschudin e0f17c75d9 ALSA: Fix assignment of 0/1 to bool variables
Convert 0 to false and 1 to true when assigning values to bool
variables. Inspired by commit 3db1cd5c05.

The simplified semantic patch that find this problem is as
follows (http://coccinelle.lip6.fr/):

@@
bool b;
@@
(
-b = 0
+b = false
|
-b = 1
+b = true
)

Signed-off-by: Peter Senna Tschudin <peter.senna@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-09-26 09:57:24 +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
Maksim A. Boyko 140d37de62 ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525
Add the volume control quirk for avoiding the kernel warning
for the Logitech HD Webcam C525
as in the similar commit 36691e1be6
for the Logitech HD Webcam C310.

Reported-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Tested-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Cc: <stable@vger.kernel.org> # 3.10.5+
Signed-off-by: Maksim Boyko <maksim.a.boyko@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-12 14:55:20 +02:00
Clemens Ladisch aa773bfe8f ALSA: usb-audio: fix automatic Roland/Yamaha MIDI detection
Commit aafe77cc45 (ALSA: usb-audio: add support for many Roland/Yamaha
devices) had several logic errors that prevented create_auto_midi_quirk
from enumerating any MIDI ports.

Reported-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-12 11:42:28 +02:00
Torsten Schenk 4c2aee0032 ALSA: 6fire: make buffers DMA-able (midi)
Patch makes midi output buffer DMA-able by allocating it separately.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-12 11:42:28 +02:00
Torsten Schenk 5ece263f1d ALSA: 6fire: make buffers DMA-able (pcm)
Patch makes pcm buffers DMA-able by allocating each one separately.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-12 11:42:28 +02:00
Andy Shevchenko 663819fb7d ALSA: don't push static constants on stack for %*ph
There is no need to pass constants via stack. The width may be explicitly
specified in the format.

Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-08 12:04:18 +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
Jussi Kivilinna ddb6b5a964 ALSA: 6fire: fix DMA issues with URB transfer_buffer usage
Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to
be DMA-able, which stack is not. Furthermore, transfer_buffer should not be
allocated as part of larger device structure because DMA coherency issues and
patch fixes this issue too.

Cc: stable@vger.kernel.org
Signed-off-by: Jussi Kivilinna <jussi.kivilinna@iki.fi>
Tested-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-07 16:04:27 +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
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
Dan Carpenter 85054b2153 ALSA: usx2y: remove an unneeded check
The test here is always true because S[i].urb is an array not a pointer.
Also it's bogus because the intent was to test:
	if (S->urb[i]) {
instead of:
	if (S[i].urb) {

Anyway, usb_kill_urb() and usb_free_urb() accept NULL pointers so we can
just remove this.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-29 13:59:47 +02:00
Eldad Zack fee4b700a4 ALSA: hiface: return correct XRUN indication
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of hiface, as expected by snd_pcm_update_hw_ptr0().

Caught by sparse.

Cc: Antonio Ospite <ospite@studenti.unina.it>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-24 10:51:37 +02:00
Eldad Zack be2f93a4c4 ALSA: usb-audio: 6fire: return correct XRUN indication
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of 6fire, as expected by snd_pcm_update_hw_ptr0().

Caught by sparse.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-21 11:55:22 +02:00
Takashi Iwai 5be1efb4c2 ALSA: usx2y: Fix unlocked snd_pcm_stop() call
snd_pcm_stop() must be called in the PCM substream lock context.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15 21:25:13 +02:00
Takashi Iwai 9538aa46c2 ALSA: ua101: Fix unlocked snd_pcm_stop() call
snd_pcm_stop() must be called in the PCM substream lock context.

Cc: <stable@vger.kernel.org>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15 21:24:57 +02:00
Takashi Iwai 5b9ab3f732 ALSA: 6fire: Fix unlocked snd_pcm_stop() call
snd_pcm_stop() must be called in the PCM substream lock context.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15 18:12:50 +02:00
Eldad Zack 42d4ab832d ALSA: usb-audio: fix regression for fixed stream quirk
Commit 8f898e92ae removed the redundant
reads of bInterfaceProtocol from the descriptors, but introduced a
regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT,
since fp->protocol is not set in setup process.

As a consequence, audio streams would not get initialized, as the
following logs show:

[   48.923043] setting usb interface 3:1
[   48.923056] Creating new capture data endpoint #81
[   48.923484] 4:3:1: cannot set freq 48000 to ep 0x81

This patch sets fp->protocol in create_fixed_stream_quirk() and
resolves the regression.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-10 17:52:14 +02:00
Przemek Rudy 066624c6a1 ALSA: usb-audio: Add Audio Advantage Micro II
This patch is adding extensive support (beside standard usb audio class)
for Audio Advantage Micro II usb sound card.
Features included:
- Access to AES bits (so now sending the IEC61937 compliant stream is
possible).
- Mixer SPDIF control added to turn on/off the optical transmitter.

Signed-off-by: Przemek Rudy <prudy1@o2.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-28 13:37:12 +02:00
Takashi Iwai ea70ee057c Merge branch 'full-roland-support' of git://git.alsa-project.org/alsa-kprivate into for-next
For adding support for many Roland and Yamaha devices:
* 'full-roland-support' of git://git.alsa-project.org/alsa-kprivate:
  ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
  ALSA: usb-audio: claim autodetected PCM interfaces all at once
  ALSA: usb-audio: remove superfluous Roland quirks
  ALSA: usb-audio: add MIDI port names for some Roland devices
  ALSA: usb-audio: add support for many Roland/Yamaha devices
  ALSA: usb-audio: detect implicit feedback on Roland devices
  ALSA: usb-audio: store protocol version in struct audioformat
2013-06-28 12:13:26 +02:00
Clemens Ladisch b7f33917bc ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
The Roland Quad/Octo-Capture devices use some unknown vendor-specific
mechanism to switch sample rates (and to manage other controls).  To
prevent the driver from attempting to use any other than the default
44.1 kHz sample rate, use quirks to hide the other alternate settings.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:50 +02:00
Clemens Ladisch b1ce7ba619 ALSA: usb-audio: claim autodetected PCM interfaces all at once
snd_card_register() registers all devices newly added since the last
call.  However, the playback/capture streams are handled as one ALSA
device, so the second /dev device will not be registered if the PCM
streams are added in two steps.

QUIRK_AUTODETECT caused the probe callback to be called once for each
interface, which triggered this problem.  Work around this by handling
this like the composite quirk, i.e., autodetecting all other interfaces
that might be used for PCM or MIDI.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:49 +02:00
Clemens Ladisch 8e5ced83dd ALSA: usb-audio: remove superfluous Roland quirks
Remove all quirks that are no longer needed now that the generic Roland
quirks can handle the vendor-specific descriptors correctly.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:49 +02:00
Clemens Ladisch a968782e27 ALSA: usb-audio: add MIDI port names for some Roland devices
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:48 +02:00
Clemens Ladisch aafe77cc45 ALSA: usb-audio: add support for many Roland/Yamaha devices
Add quirks to detect the various vendor-specific descriptors used by
Roland and Yamaha in most of their recent USB audio and MIDI devices.

Together with the previous patch, this should add audio/MIDI support for
the following USB devices:
- Edirol motion dive .tokyo performance package
- Roland MC-808 Synthesizer
- Roland BK-7m Synthesizer
- Roland VIMA JM-5/8 Synthesizer
- Roland SP-555 Sequencer
- Roland V-Synth GT Synthesizer
- Roland Music Atelier AT-75/100/300/350C/500/800/900/900C Organ
- Edirol V-Mixer M-200i/300/380/400/480/R-1000
- BOSS GT-10B Effects Processor
- Roland Fantom G6/G7/G8 Keyboard
- Cakewalk Sonar V-Studio 20/100/700 Audio Interface
- Roland GW-8 Keyboard
- Roland AX-Synth Keyboard
- Roland JUNO-Di/STAGE/Gi Keyboard
- Roland VB-99 Effects Processor
- Cakewalk UM-2G MIDI Interface
- Roland A-500S Keyboard
- Roland SD-50 Synthesizer
- Roland OCTAPAD SPD-30 Controller
- Roland Lucina AX-09 Synthesizer
- BOSS BR-800 Digital Recorder
- Roland DUO/TRI-CAPTURE (EX) Audio Interface
- BOSS RC-300 Loop Station
- Roland JUPITER-50/80 Keyboard
- Roland R-26 Recorder
- Roland SPD-SX Controller
- BOSS JS-10 Audio Player
- Roland TD-11/15/30 Drum Module
- Roland A-49/88 Keyboard
- Roland INTEGRA-7 Synthesizer
- Roland R-88 Recorder

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:48 +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
Antonio Ospite a91c3fb2f8 Add M2Tech hiFace USB-SPDIF driver
Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices.

M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output
Interface, see http://www.m2tech.biz/hiface.html

The supported products are:

  * M2Tech Young
  * M2Tech hiFace
  * M2Tech North Star
  * M2Tech W4S Young
  * M2Tech Corrson
  * M2Tech AUDIA
  * M2Tech SL Audio
  * M2Tech Empirical
  * M2Tech Rockna
  * M2Tech Pathos
  * M2Tech Metronome
  * M2Tech CAD
  * M2Tech Audio Esclusive
  * M2Tech Rotel
  * M2Tech Eeaudio
  * The Chord Company CHORD
  * AVA Group A/S Vitus

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-24 09:26:08 +02:00
Antonio Ospite 0af49ffe3c ALSA: usb: uniform style used in MODULE_SUPPORTED_DEVICE()
In sound/usb/card.c and sound/usb/misc/ua101.c there are no spaces
between the vendor and the device names, use this style in the other
drivers too.

This also helps keeping consistency when new drivers copies from the
ones already in the mainline tree.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:37:08 +02:00
Antonio Ospite 4a9f911861 ALSA: snd-usb-6fire: use vmalloc buffers
For USB devices it's not necessary to allocate physically contiguous
buffers.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:36:41 +02:00
Antonio Ospite fc76f86376 ALSA: snd-usb-caiaq: use vmalloc buffers
For USB devices it's not necessary to allocate physically contiguous
buffers.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:35:52 +02:00
Antonio Ospite 3dd446a7e5 ALSA: snd-usb-caiaq: remove the unused snd_card_used variable
The snd_card_used variable is only read but never written, remove it.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:33:05 +02:00
Dave Jones cd1199edc7 ALSA: sound/usb/misc/ua101.c: convert __list_for_each usage to list_for_each
Signed-off-by: Dave Jones <davej@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-18 07:47:32 +02:00
Dan Carpenter da177dd025 ALSA: usx2y: remove some old dead code
USB_QUEUE_BULK isn't defined any more.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17 10:45:42 +02:00
Takashi Iwai 36691e1be6 ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310
Just like the previous fix for LogitechHD Webcam c270 in commit
11e7064f35, c310 model also requires the
same workaround for avoiding the kernel warning.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59741
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17 10:25:02 +02:00
Clemens Ladisch 342cda2934 ALSA: usb-audio: work around Android accessory firmware bug
When the Android firmware enables the audio interfaces in accessory
mode, it always declares in the control interface's baInterfaceNr array
that interfaces 0 and 1 belong to the audio function.  However, the
accessory interface itself, if also enabled, already is at index 0 and
shifts the actual audio interface numbers to 1 and 2, which prevents the
PCM streaming interface from being seen by the host driver.

To get the PCM interface interface to work, detect when the descriptors
point to the (for this driver useless) accessory interface, and redirect
to the correct one.

Reported-by: Jeremy Rosen <jeremy.rosen@openwide.fr>
Tested-by: Jeremy Rosen <jeremy.rosen@openwide.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17 09:56:52 +02:00
Takashi Iwai 11e7064f35 ALSA: usb-audio - Fix invalid volume resolution on Logitech HD webcam c270
USB audio driver spews an error message when probing Logitech HD
webcam c270:
  ALSA mixer.c:1300 usb_audio: Warning! Unlikely big volume range (=6144), cval->res is probably wrong.
  ALSA mixer.c:1304 usb_audio: [5] FU [Mic Capture Volume] ch = 1, val = 1536/7680/1

Obviously the device needs a fixed volume resolution (cval->res = 384)
like other Logitech devices.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=821735

Reported-and-tested-by: Cristian Rodríguez <crrodriguez@opensuse.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-05 08:35:26 +02:00
Takashi Iwai 8eafc0a161 ALSA: usb-audio - Apply Logitech QuickCam Pro 9000 quirk only to audio iface
... instead of applying to all interfaces.

Reference: http://forums.gentoo.org/viewtopic-p-6886404.html

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-04 16:07:48 +02:00
Clemens Ladisch a0c6d309c6 ALSA: usb-audio: fix Roland/Cakewalk UM-3G support
Commit 927c9423dd (ALSA: usb-audio: add
Edirol UM-3G support) used a wrong quirk type, which would make the
driver refuse to attach with the error message "MIDIStreaming interface
descriptor not found".

Cc: <stable@vger.kernel.org> # 3.3 and later
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-03 09:42:21 +02:00
Torsten Schenk d47333ddb2 ALSA: usb-6fire: Modify firmware version check
Check only the uppermost 16 bits instead of the whole 32 bits of
the version information. Do this because all firmware version tested
with this version information worked correctly and the strict check
causes problems for several users.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-05-23 14:30:26 +02:00
Torstein Hegge e6135fe960 ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format
freqshift is only set for the data endpoint and syncmaxsize is only set
for the sync endpoint. This results in a syncmaxsize of zero used in the
proc output feedback format calculation, which gives a feedback format
incorrectly shown as 8.16 for UAC2 devices.

As neither the data nor the sync endpoint gives all the relevant
content, output the two combined.

Also remove the sync_endpoint "packet size" which is always zero
and the sync_endpoint "momentary freq" which is constant.

Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async.

Reported-by: B. Zhang <bb.zhang@free.fr>
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-05-17 08:05:34 +02:00
Eldad Zack 4ca231b2e6 ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
Current code does this:

  be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1])

Which is effectively (neglecting the index):

  be16_to_cpu(be16_to_cpu(*((u16 *) buf)))

This means the int16 in the buffer is not converted at all.

Daniel Mack confirmed that the driver works on little endian
CPUs, leading to the conclusion that the device-side structure
is actually little endian.
This changes the code to use le16_to_cpu().

Caught by sparse.

Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-30 09:19:02 +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
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
David Henningsson fa92dd77ec ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
The Scarlett 2i2 seems to take almost 500 ms to set the sample rate,
even if the clock is currently set to that value. This patch speeds
up prepare of the device, by avoiding setting the clock to something
it already is.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-26 07:37:09 +02:00
Trulan Martin 03e0221444 ALSA: usb-audio: USB quirk for Yamaha THR10C
This patch adds a USB quirk for the Yamaha THR10C amp.

Signed-off-by: Trulan Martin <trulanm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:48:21 +02:00
Trulan Martin 1b15362c74 ALSA: usb-audio: USB quirk for Yamaha THR5A
This patch adds a USB quirk for the Yamaha THR5A amp.

Signed-off-by: Trulan Martin <trulanm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:48:02 +02:00
Trulan Martin ae3f0c267f ALSA: usb-audio: USB quirk for Yamaha THR10
This patch adds a USB quirk for the Yamaha THR10 amp.

Signed-off-by: Trulan Martin <trulanm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:47:50 +02:00
Takashi Iwai 60af3d037e ALSA: usb-audio: Fix autopm error during probing
We've got strange errors in get_ctl_value() in mixer.c during
probing, e.g. on Hercules RMX2 DJ Controller:

  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4
  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4
  ....

It turned out that the culprit is autopm: snd_usb_autoresume() returns
-ENODEV when called during card->probing = 1.

Since the call itself during card->probing = 1 is valid, let's fix the
return value of snd_usb_autoresume() as success.

Reported-and-tested-by: Daniel Schürmann <daschuer@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:46:51 +02:00
Daniel Mack ebfc594c02 ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.

There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.

When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.

Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Torstein Hegge <hegge@resisty.net>
Reported-and-tested-by: Yves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:33:20 +02:00
Daniel Schürmann b5f035dbca ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
Set the timeout for USB control set messages according to the USB 2
spec, using the macros from include/linux/usb.h.
The get timout becomes 5000 ms even though it is 500 ms in the
spec. This patch is required to run the Hercules RMX2 which needs a
timeout of 1240 ms.

More notes from author:
I still distinguish between set and get but as long both are 5000 ms
GCC will remove it anyway. IMHO this is more easy read and there is no
need to explain why we use a get timeout for set messages.

Signed-off-by: Daniel Schürmann <daschuer@mixxx.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-22 10:45:02 +02:00
Takashi Iwai 8dd2b66d1a ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
 platform conversions which have been tested - getting this in mainline
 will make life easier for development after the merge window.  These
 factor a large chunk of code out of the drivers for the platforms using
 dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: More updates for v3.10

The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window.  These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
2013-04-18 16:24:31 +02:00
Daniel Mack 126825e7ea ALSA: snd-usb: add quirks handler for DSD streams
Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.

That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.

The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:53 +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
Clemens Ladisch cbc200bca4 ALSA: usb-audio: disable autopm for MIDI devices
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices.  However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions.  With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.

Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.

Commit f5f165418c (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.

To work around all this, just disable autopm for all USB MIDI devices.

Reported-by: Laurens Holst
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-15 16:03:57 +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 889d66848b ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()

Caught by sparse:

sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38:    got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35:    got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56:    got restricted __le16 [usertype] <noident>
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35:    got restricted __le16 [usertype] <noident>

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-07 09:44:08 +02:00
Eldad Zack 1dc669fed6 ALSA: usb-audio: UAC2: support read-only freq control
Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).

In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.

If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:32:07 +02:00
Eldad Zack 027bbc1546 ALSA: usb-audio: show err in set_sample_rate_v2 debug
Show the error code returned from the USB subsystem in
the debug messages.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:31:40 +02:00
Eldad Zack ef02e29b01 ALSA: usb-audio: UAC2: auto clock selection module param
Add a module param to disable auto clock selection.
This is provided for users that expect the audio stream to
fail when the clock source is invalid (e.g., the word clock
was unintentionally disconnected).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:31:32 +02:00
Eldad Zack 8c55af3f69 ALSA: usb-audio: UAC2: try to find and switch to valid clock
If a selector is available on a device, it may be pointing to a
clock source which is currently invalid.
If there is a valid clock source which can be selected, switch
to it.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:31:14 +02:00
Eldad Zack 06ffc1ebdd ALSA: usb-audio: UAC2: do clock validity check earlier
Move the check that parse_audio_format_rates_v2() do after
receiving the clock source entity ID directly into the find
function and add a validation flag to the function.

This patch does not introduce any logic flow change.

It is provided to allow introducing automatic clock switching
easier later. By moving this uac_clock_source_is_valid callsite,
2 additional callsites can be avoided.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:30:59 +02:00
Eldad Zack f6a8bc70f8 ALSA: usb-audio: use endianness macros
Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:30:49 +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 ed136aca77 ALSA: usb-audio: neaten EXPORT_SYMBOLS placement
Put EXPORT_SYMBOLS directly under the exported function.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:30:24 +02:00
Eldad Zack f9d3543591 ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placement
Minor style fix, following a general code style in the kernel.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:30:18 +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
Takashi Iwai 7c51746517 ALSA: usb-audio: Clean up the code in set_sample_rate_v2()
Just for cleaning up, introduce a new function get_sample_rate_v2()
for replacing two identical calls in set_sample_rate_v2().

No functional change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-03 19:08:29 +02:00
Takashi Iwai efc33ce197 Merge branch 'for-linus' into for-next
Back-merge for cleaning up usb-audio code the recent commit modified,
and further UAC2 autoclock patches.
2013-04-03 17:07:29 +02:00
Torstein Hegge 690a863ff0 ALSA: usb: Work around CM6631 sample rate change bug
The C-Media CM6631 USB receiver doesn't respond to changes in sample rate
while the interface is active. The same behavior is observed in other UAC2
hardware like the VIA VT1731.

Reset the interface after setting the sampling frequency on sample rate
changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is
used. Otherwise, the device will try to use the sample rate of the previous
stream, causing distorted sound on sample rate changes.

The reset is performed for all UAC2 devices, as it should not affect a
standards compliant device, but it is only necessary for C-Media CM6631,
VIA VT1731 and possibly others.

Failure to read sample rate from the device is not handled as an error in
set_sample_rate_v2(), as (permanent or intermittent) failure to read sample
rate isn't essential for a successful sample rate set.

Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-03 17:05:44 +02:00
Takashi Iwai 10d7410790 Merge branch 'for-linus' into for-next
Merge back for-linus branch for the badness table adjustment for VIA codecs

* for-linus:
  ALSA: hda - Fix DAC assignment for independent HP
  ALSA: hda - Fix abuse of snd_hda_lock_devices() for DSP loader
  ALSA: hda - Fix typo in checking IEC958 emphasis bit
  ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
  ALSA: snd-usb: mixer: propagate errors up the call chain
  ALSA: usb: Parse UAC2 extension unit like for UAC1
  ALSA: hda - Fix yet missing GPIO/EAPD setup in cirrus driver
2013-03-22 14:53:25 +01:00
Daniel Mack 83ea5d18d7 ALSA: snd-usb: mixer: ignore -EINVAL in snd_usb_mixer_controls()
Creation of individual mixer controls may fail, but that shouldn't cause
the entire mixer creation to fail. Even worse, if the mixer creation
fails, that will error out the entire device probing.

All the functions called by parse_audio_unit() should return -EINVAL if
they find descriptors that are unsupported or believed to be malformed,
so we can safely handle this error code as a non-fatal condition in
snd_usb_mixer_controls().

That fixes a long standing bug which is commonly worked around by
adding quirks which make the driver ignore entire interfaces. Some of
them might now be unnecessary.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Rodolfo Thomazelli <pe.soberbo@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20 08:43:00 +01:00
Daniel Mack 4d7b86c98e ALSA: snd-usb: mixer: propagate errors up the call chain
In check_input_term() and parse_audio_feature_unit(), propagate the
error value that has been returned by a failing function instead of
-EINVAL. That helps cleaning up the error pathes in the mixer.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20 08:42:35 +01:00
Torstein Hegge 61ac51301e ALSA: usb: Parse UAC2 extension unit like for UAC1
UAC2_EXTENSION_UNIT_V2 differs from UAC1_EXTENSION_UNIT, but can be handled in
the same way when parsing the unit. Otherwise parse_audio_unit() fails when it
sees an extension unit on a UAC2 device.

UAC2_EXTENSION_UNIT_V2 is outside the range allocated by UAC1.

Signed-off-by: Torstein Hegge <hegge@resisty.net>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-20 08:42:12 +01:00
Takashi Iwai cf30f46acd Merge branch 'for-linus' into for-next
Back-merged for refactoring beep stuff.
2013-03-18 11:04:42 +01:00
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
Daniel Mack 717bfb5f46 ALSA: snd-usb: handle raw data format of UAC2 devices
UAC2 compliant audio devices may announce the capability to transport
raw audio data on their endpoints. Catch this and handle it as
'special' stream on the ALSA side.

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 08:47:13 +01:00
Daniel Mack 2fcdb06d49 ALSA: snd-usb: handle the bmFormats field as unsigned int
This field may use up to 32 bits, so it should be handled as unsigned
int.

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 08:47:04 +01:00
Mark Hills 59ea586f54 ALSA: usb-audio: Trust fields given in the quirk
The maxpacksize field is given in some quirks, but it gets ignored (in
favour of wMaxPacketSize from the first endpoint.) This patch favours
the one in the quirk.

Digidesign Mbox and Mbox 2 are the only affected quirks and the devices
are assumed to be working without this patch. So for safety against the
values in the quirk being incorrect, remove them.

The datainterval is also ignored but there are not currently any quirks
which choose to override this.

Cc: Damien Zammit <damien@zamaudio.com>
Cc: Chris J Arges <christopherarges@gmail.com>
Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:46:37 +01:00
Mark Hills 5e212332cc ALSA: usb-audio: Playback and MIDI support for Novation Twitch DJ controller
The hardware also has a PCM capture device which is not implemented in
this patch.

It may be possible to generalise this to Saffire 6 USB support and some
of the other Focusrite interfaces, but as I don't have access to these
devices we should wait until capture support is working first.

Capture support is not implemented because the code assumes the endpoint
to have its own interface (instead, it shares the interface with playback)
and some thought will be needed to lift this limitation.

Signed-off-by: Mark Hills <mark@xwax.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-18 08:46:18 +01:00
Clemens Ladisch 281a6ac0f5 ALSA: usb-audio: add a workaround for the NuForce UDH-100
The NuForce UDH-100 numbers its interfaces incorrectly, which makes the
interface associations come out wrong, which results in the driver
erroring out with the message "Audio class v2 interfaces need an
interface association".

Work around this by searching for the interface association descriptor
also in some other place where it might have ended up.

Reported-and-tested-by: Dave Helstroom <helstroom@google.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-12 08:35:30 +01:00
Daniel Mack 2dad940219 ALSA: snd-usb-caiaq: fix smatch warnings
Fix three smatch warnings recently introduced:

sound/usb/caiaq/device.c:166 usb_ep1_command_reply_dispatch() warn:
  variable dereferenced before check 'cdev' (see line 163)
sound/usb/caiaq/device.c:517 snd_disconnect() warn: variable
  dereferenced before check 'card' (see line 514)
sound/usb/caiaq/input.c:510 snd_usb_caiaq_ep4_reply_dispatch() warn:
  variable dereferenced before check 'cdev' (see line 506)

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-07 09:24:12 +01:00
Daniel Mack f1f6b8f65f ALSA: snd-usb-caiaq: switch to dev_*() logging
Get rid of the proprietary functions log() and debug() and use the
generic dev_*() approach. A macro is needed to cast a cdev to a struct
device *.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-04 09:57:26 +01:00
Daniel Mack 1c8470ce31 ALSA: snd-usb-caiaq: rename 'dev' to 'cdev'
This is needed in order to make the device namespace cleaner, and will
help when moving this driver over to dev_*() logging.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-03-04 09:57:17 +01:00
Jiri Slaby 4909a0caab ALSA: usb/quirks, fix out-of-bounds access
bootresponse in snd_usb_mbox2_boot_quirk is only 12 (decimal) u8's
long, but i9s passed to snd_usb_ctl_msg as it would be 0x12 (hexa)
long. Fix that by having proper size of the array, i.e. 0x12.

Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-02-17 18:02:00 +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 2faea5274f Merge branch 'for-linus' into for-next
Merge pending fixes that haven't pulled into 3.8.
2013-02-05 14:48:03 +01:00
Takashi Iwai 8058e14259 Merge branch 'usb-audio-fix' of git://git.alsa-project.org/alsa-kprivate into for-linus 2013-02-01 07:22:47 +01:00
Clemens Ladisch 7da5804648 ALSA: usb-audio: fix Roland A-PRO support
The quirk for the Roland/Cakewalk A-PRO keyboards accidentally used the
wrong interface number, which prevented the driver from attaching to the
device.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.37+ <stable@vger.kernel.org>
2013-01-31 21:21:59 +01:00
Antonio Ospite aa53f98674 ALSA: usb: cosmetics, remove a leading space
Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-29 15:11:13 +01:00
Antonio Ospite febd1cc438 ALSA: caiaq: fix use of MODULE_SUPPORTED_DEVICES()
It looks like MODULE_SUPPORTED_DEVICES() is not implemented yet, but
still, having the entries in the list consistently separated by commas
and with balanced parenthesis won't hurt.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-29 15:10:57 +01:00
Clemens Ladisch d56268fb10 ALSA: usb-audio: fix invalid length check for RME and other UAC 2 devices
Commit 23caaf19b1 (ALSA: usb-mixer: Add support for Audio Class v2.0)
forgot to adjust the length check for UAC 2.0 feature unit descriptors.
This would make the code abort on encountering a feature unit without
per-channel controls, and thus prevented the driver to work with any
device having such a unit, such as the RME Babyface or Fireface UCX.

Reported-by: Florian Hanisch <fhanisch@uni-potsdam.de>
Tested-by: Matthew Robbetts <wingfeathera@gmail.com>
Tested-by: Michael Beer <beerml@sigma6audio.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: 2.6.35+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-27 10:22:56 +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
Eldad Zack 39e95156b9 ALSA: usb-audio: selector map for M-Audio FT C400
Add names of the clock sources for the M-Audio Fast Track
C400.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:11 +01:00
Eldad Zack 83e3acd494 ALSA: usb-audio: M-Audio FT C400 skip packet quirk
Attain constant real-world latency by skipping 16 data packets.
The number of packets to be skipped was found by trial and error.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:06:03 +01:00
Eldad Zack 2aad272b3f ALSA: usb-audio: correct M-Audio C400 clock source quirk
Taking another look at the C400 descriptors, I see now that there is
a clock selector (0x80) for this device.
Right now, the clock source points to the internal clock (0x81), which
is also valid. When the external clock source (0x82) is selected in the
mixer, and the rates mismatch (if it's free-running it is fixed to
48KHz), xruns will occur.

Set the clock ID to the clock selector unit (0x81), which then
allows the validation code to function correctly.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:05:57 +01:00
David Henningsson b98ae2729d ALSA: usb - fix race in creation of M-Audio Fast track pro driver
A patch in the 3.2 kernel caused regression with hotplugging the
M-Audio Fast track pro, or sound after suspend. I don't have the
device so I haven't done a full analysis, but it seems userspace
(both udev and pulseaudio) got confused when a card was created,
immediately destroyed, and then created again.

However, at least one person in the bug report (martin djfun)
reports that this patch resolves the issue for him. It also leaves
a message in the log:
"snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is
a bit misleading. It is better than non-working audio, but maybe
there's a more elegant solution?

BugLink: https://bugs.launchpad.net/bugs/1095315
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-14 10:03:03 +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
Sachin Kamat e8e7da23c9 ALSA: usb-audio: Make ebox44_table static
Fixes the following sparse warning:
sound/usb/mixer_quirks.c:1209:23: warning:
symbol 'ebox44_table' was not declared. Should it be static?

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:22:25 +01:00
Damien Zammit b7b435e81b ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirk
This patch is based on 3.8-rc1. It fixes two things:
1) A kernel panic caused by incorrect allocation of a u8 variable
   "bootresponse".
2) A noisy dmesg (urb status -32) caused by broken pipe to an
   invalid midi endpoint.

It is also a little cleaner because there is no need for a new
QUIRK_MIDI type as suggested by kernel developers, since the device
follows exactly the MIDIMAN protocol.

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 09:53:17 +01:00
Alexander Schremmer 8f7f3ab15e ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirks
Support the Creative BT-D1 Bluetooth USB audio device. Before this
patch, Linux had trouble finding the correct USB descriptors and bailed
out with these messages:

 no or invalid class specific endpoint descriptor

Now it still prints these messages on hotplug:

 snd-usb-audio: probe of ...:1.0 failed with error -5
 snd-usb-audio: probe of ...:1.2 failed with error -5
 snd-usb-audio: probe of ...:1.3 failed with error -5

But the device works correctly, including the HID support.

The patch is diff'ed against 3.8-rc1 but should apply to older kernels
as well.

Signed-off-by: Alexander Schremmer <alex@alexanderweb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:26:48 +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
Linus Torvalds 03c850ec32 Sound fixes for 3.8-rc1
This update contains overall only driver-specific fixes.
 Slightly large LOC are seen in usb-audio driver for a couple of new
 device quirks and cs42l71 ASoC driver for enhanced features.
 The others are a few small (regression) fixes HD-audio, and yet other
 small / trival ASoC fixes.
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Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This update contains overall only driver-specific fixes.  Slightly
  large LOC are seen in usb-audio driver for a couple of new device
  quirks and cs42l71 ASoC driver for enhanced features.  The others are
  a few small (regression) fixes HD-audio, and yet other small / trival
  ASoC fixes."

* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
  ALSA: HDA: Fix sound resume hang
  ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
  ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
  ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
  ASoC: atmel-ssc: change disable to disable in dts node
  ASoC: Prevent pop_wait overwrite
  ALSA: usb-audio: ignore-quirk for HP Wireless Audio
  ALSA: hda - Always turn on pins for HDMI/DP
  ALSA: hda - Fix pin configuration of HP Pavilion dv7
  ASoC: core: Fix splitting of log messages
  ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
  ASoC: cs42l73: Add DAPM events for power down.
  ASoC: cs42l73: Add DMIC's as DAPM inputs.
  ASoC: sigmadsp: Fix endianness conversion issue
  ASoC: tpa6130a2: Use devm_* APIs
2012-12-20 07:52:13 -08:00
Damien Zammit cb99864d40 ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
This patch is the result of a lot of trial and error, since there are no specs
available for the device.

Full duplex support is provided, i.e. playback and recording in stereo.
The format is hardcoded at 48000Hz @ 24 bit, which is the maximum that the
device supports.  Also, MIDI in and MIDI out both work.

Users will notice that the S/PDIF light also flashes when playback or recording
is active.  I believe this means that S/PDIF input/output is simultaneously
activated with the analogue i/o during use.
But this particular functionality remains untested.

Note that this particular version of the patch is so far untested on the
physical hardware because I have not compiled a full kernel with the changes.
However, extensive testing has been done by many users of the hardware
who believe other versions of my patch have worked since circa 2009.

[Modified to make a function static by tiwai]

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-19 11:27:22 +01:00
Eldad Zack df68f10643 ALSA: usb-audio: ignore-quirk for HP Wireless Audio
As Joe Cooper <swelljoe@gmail.com> reported, "On most HP Envy laptops
the snd-usb-audio module causes the system to become unresponsive and
Gnome Shell 3 to crash.".
See also:
 http://mailman.alsa-project.org/pipermail/alsa-devel/2012-December/057729.html

Add a quirk to ignore this device (for now) to solve the instability
issue and allow other USB audio devices to be used.

Reported-by: Joe Cooper <swelljoe@gmail.com>
Tested-by: Isaac Smith <hunternet93@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-15 11:13:10 +01:00
Linus Torvalds a2013a13e6 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial
Pull trivial branch from Jiri Kosina:
 "Usual stuff -- comment/printk typo fixes, documentation updates, dead
  code elimination."

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (39 commits)
  HOWTO: fix double words typo
  x86 mtrr: fix comment typo in mtrr_bp_init
  propagate name change to comments in kernel source
  doc: Update the name of profiling based on sysfs
  treewide: Fix typos in various drivers
  treewide: Fix typos in various Kconfig
  wireless: mwifiex: Fix typo in wireless/mwifiex driver
  messages: i2o: Fix typo in messages/i2o
  scripts/kernel-doc: check that non-void fcts describe their return value
  Kernel-doc: Convention: Use a "Return" section to describe return values
  radeon: Fix typo and copy/paste error in comments
  doc: Remove unnecessary declarations from Documentation/accounting/getdelays.c
  various: Fix spelling of "asynchronous" in comments.
  Fix misspellings of "whether" in comments.
  eisa: Fix spelling of "asynchronous".
  various: Fix spelling of "registered" in comments.
  doc: fix quite a few typos within Documentation
  target: iscsi: fix comment typos in target/iscsi drivers
  treewide: fix typo of "suport" in various comments and Kconfig
  treewide: fix typo of "suppport" in various comments
  ...
2012-12-13 12:00:02 -08:00
Denis Washington 1d31affbef ALSA: usb-audio: Enable S/PDIF on the ASUS Xonar U3
The only required change is to extend the existing Xonar U1
mixer quirks to the U3, which seems to be controlled the same
way.

Signed-off-by: Denis Washington <denisw@online.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-12 11:32:54 +01:00
Jurgen Kramer 9621055fbb ALSA: usb6fire: prevent driver panic state when stopping
The patch below prevents the 6fire usb driver going into panic state
when stopping playing. On some systems the urb in handler
(usb6fire_pcm_in_urb_handler) is being called while urbs are being
killed off, this causes the driver to set panic state and can result in
the kernel warning 'URB %p submitted while active'.

Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-07 15:03:34 +01:00
Bill Pemberton 14c56706f9 ALSA: snd-usb-caiaq: remove __dev* attributes
CONFIG_HOTPLUG is going away as an option.  As result the __dev*
markings will be going away.

Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.

Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-07 07:35:11 +01:00
Bill Pemberton 87f9796a03 ALSA: snd-usb-6fire: remove __dev* attributes
CONFIG_HOTPLUG is going away as an option.  As result the __dev*
markings will be going away.

Remove use of __devinit, __devexit_p, __devinitdata, __devinitconst,
and __devexit.

Signed-off-by: Bill Pemberton <wfp5p@virginia.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-07 07:34:46 +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
Takashi Iwai f5f165418c ALSA: usb-audio: Fix missing autopm for MIDI input
The commit [88a8516a: ALSA: usbaudio: implement USB autosuspend] added
the support of autopm for USB MIDI output, but it didn't take the MIDI
input into account.

This patch adds the following for fixing the autopm:
- Manage the URB start at the first MIDI input stream open, instead of
  the time of instance creation
- Move autopm code to the common substream_open()
- Make snd_usbmidi_input_start/_stop() more robust and add the running
  state check

Reviewd-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04 07:27:44 +01:00
Takashi Iwai 59866da9e4 ALSA: usb-audio: Avoid autopm calls after disconnection
Add a similar protection against the disconnection race and the
invalid use of usb instance after disconnection, as well as we've done
for the USB audio PCM.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=51201

Reviewd-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-04 07:27:27 +01:00
David Henningsson 9b4ef97757 ALSA: usb - Don't create "Speaker" mixer controls on headphones and headsets
A lot of headsets/headphones have a "Speaker" mixer control. This confuses
PulseAudio to think it is a speaker instead of a headphone/headset.
Therfore, we rename it to "Headphone".

We determine if something is a headphone similar to how udev determines
form factor (see 78-sound-card.rules).

BugLink: https://bugs.launchpad.net/bugs/1082357
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 13:59:47 +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 09d8e3a71d ALSA: usb-audio: Fast Track C400 mixer controls
Add a mixer quirks for the M-Audio Fast Track C400
and create the following:

* Volume controls
* Effect Type (reusing FTU controls)
* Effect Volume
* Effect Send/Return
* Effect Program
* Effect Feedback

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:44:55 +01:00
Eldad Zack d50ed624e4 ALSA: usb-audio: Fast Track C400 mixer ranges
Add ranges for various Fast Track C400 controls, as observed
while using the vendor's mixer control software (res values
are an estimation).

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:44:42 +01:00
Eldad Zack 76f74bca73 ALSA: usb-audio: M-Audio Fast Track C400 quirks table
Adds a quirks table for the M-Audio Fast Track C400.
Thanks to Clemens Ladisch <clemens@ladisch.de> for pointing out that
the table must be sorted.

Based on the following patch from the alsa-devel list:
http://mailman.alsa-project.org/pipermail/alsa-devel/2012-May/051676.html

See also:
http://mailman.alsa-project.org/pipermail/alsa-devel/2012-April/051219.html

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:44:17 +01:00
Eldad Zack d847ce0e9a ALSA: usb-audio: parameterize FTU effect unit control
Adds the unit ID and the control as parameters to the creation of the
effect unit control for the M-Audio Fast Track Ultra. This allows the
code to be shared with other devices that use different unit ID and
control, such as the M-Audio Fast Track C400.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:43:55 +01:00
Eldad Zack 5dae5fd240 ALSA: usb-audio: skip UAC2 EFFECT_UNIT
Current code mishandles the case where the device is a UAC2
and the bDescriptorSubtype is a UAC2 Effect Unit (0x07).
It tries to parse it as a Processing Unit (which is similar to two
other UAC1 units with overlapping subtypes), but since the structure
is different (See: 4.7.2.10, 4.7.2.11 in UAC2 standard), the parsing
is done incorrectly and prevents the device from initializing.
For now, just ignore the unit.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:43:31 +01:00
Eldad Zack 9f81410592 ALSA: usb-audio: add control index offset
Currently, channel IDs exceeding 31 (0x1f) cannot be used.
The channel ID is derived from the cmask. Extending cmask
to a 64-bit type would only allow it to go up to 63 (0x3f).
Some devices have channel IDs exceeding that as well.
To address that, add an offset to the mixer element which
is then accounted for in the UAC set/get functions.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-29 08:43:12 +01:00
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
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 04324ccc75 ALSA: usb-audio: add channel map support
Add the support for channel maps of the PCM streams on USB audio
devices.  The channel map information is already found in
ChannelConfig descriptor entries, which haven't been referred until
now.

Each chmap entry is added to audioformat list entry and copied to TLV
dynamically instead of creating a whole chmap array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-26 16:24:02 +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 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
Sachin Kamat 8ad10dc6d3 ALSA: usb-audio: Return meaningful error codes instead of -1 in format.c
Also, silences the following smatch warning:
sound/usb/format.c:170 parse_audio_format_rates_v1() warn:
returning -1 instead of -ENOMEM is sloppy

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 11:31:52 +01:00
Sachin Kamat 27b2a22c71 ALSA: usb/6fire: Fix potential NULL pointer dereference in comm.c
'rt' was dereferenced before the NULL check.
Moved the code after the check.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-21 10:43:52 +01:00
Takashi Iwai 87af0b80c9 Merge branch 'for-linus' into for-next
Merge the recent HD-audio codec change for fixing recursive suspend
calls.

Conflicts:
	sound/pci/hda/hda_codec.c
2012-11-19 21:25:27 +01:00
Adam Buchbinder 48fc7f7e78 Fix misspellings of "whether" in comments.
"Whether" is misspelled in various comments across the tree; this
fixes them. No code changes.

Signed-off-by: Adam Buchbinder <adam.buchbinder@gmail.com>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
2012-11-19 14:31:35 +01:00
Takashi Iwai 0ced14fbda Merge branch 'usb-midi-fix-3.7' of git://git.alsa-project.org/alsa-kprivate into for-linus
Merge a regression fix for USB MIDI on non-standard usb-audio drivers
by Clemens.
2012-11-19 09:55:06 +01:00
Clemens Ladisch e99ddfde6a ALSA: ua101, usx2y: fix broken MIDI output
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend) added
autosuspend code to all files making up the snd-usb-audio driver.
However, midi.c is part of snd-usb-lib and is also used by other
drivers, not all of which support autosuspend.  Thus, calls to
usb_autopm_get_interface() could fail, and this unexpected error would
result in the MIDI output being completely unusable.

Make it work by ignoring the error that is expected with drivers that do
not support autosuspend.

Reported-by: Colin Fletcher <colin.m.fletcher@googlemail.com>
Reported-by: Devin Venable <venable.devin@gmail.com>
Reported-by: Dr Nick Bailey <nicholas.bailey@glasgow.ac.uk>
Reported-by: Jannis Achstetter <jannis_achstetter@web.de>
Reported-by: Rui Nuno Capela <rncbc@rncbc.org>
Cc: Oliver Neukum <oliver@neukum.org>
Cc: 2.6.39+ <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2012-11-18 17:15:24 +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 10e44239f6 ALSA: usb-audio: Fix mutex deadlock at disconnection
The recent change for USB-audio disconnection race fixes introduced a
mutex deadlock again.  There is a circular dependency between
chip->shutdown_rwsem and pcm->open_mutex, depicted like below, when a
device is opened during the disconnection operation:

A. snd_usb_audio_disconnect() ->
     card.c::register_mutex ->
       chip->shutdown_rwsem (write) ->
         snd_card_disconnect() ->
           pcm.c::register_mutex ->
             pcm->open_mutex

B. snd_pcm_open() ->
     pcm->open_mutex ->
       snd_usb_pcm_open() ->
         chip->shutdown_rwsem (read)

Since the chip->shutdown_rwsem protection in the case A is required
only for turning on the chip->shutdown flag and it doesn't have to be
taken for the whole operation, we can reduce its window in
snd_usb_audio_disconnect().

Reported-by: Jiri Slaby <jslaby@suse.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-14 15:29:09 +01:00
Martin Schwenke 1762a59d8e ALSA: usb-audio: Add quirk for Focusrite Scarlett 18i6
Probing this device currently fails in snd_usb_audio_probe() because
the call to snd_usb_create_mixer() fails.  This is due to unknown or
non-standard interface descriptor subtypes in parse_audio_unit():

  usbaudio: unit 51: unexpected type 0x09
  snd-usb-audio: probe of 1-8:1.0 failed with error -5

Some people are working around this by recompiling usb-audio with the
call to snd_usb_create_mixer() commented out.  It would be nice to
avoid that.

While the best idea would be to look into the mixer creation failure,
a reasonable short-term solution is to use quirks to only probe the
trouble-free interfaces.  This allows audio and MIDI interfaces to be
used without any obvious issues.

Interface 0 is the main one to ignore.  It contains lots of
control-fu, including the unexpected interface descriptor subtypes.
Interface 5 is for firmware updates and I'm not sure how to get
support for this.  Interface 3 is some sort of control interface that
I don't understand:

    Interface Descriptor:
      bLength                 9
      bDescriptorType         4
      bInterfaceNumber        3
      bAlternateSetting       0
      bNumEndpoints           0
      bInterfaceClass         1 Audio
      bInterfaceSubClass      1 Control Device
      bInterfaceProtocol      0
      iInterface              0
      AudioControl Interface Descriptor:
        bLength                 9
        bDescriptorType        36
        bDescriptorSubtype      1 (HEADER)
        bcdADC               1.00
        wTotalLength            9
        bInCollection           1
        baInterfaceNr( 0)       1

Signed-off-by: Martin Schwenke <martin@meltin.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-13 09:47:13 +01:00
Takashi Iwai 17a4adbe68 Merge branch 'for-linus' into for-next 2012-11-08 15:58:25 +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
Takashi Iwai a5d00dc3a4 Merge branch 'for-linus' into for-next
... for migrating the core changes for USB-audio disconnection fixes
2012-10-30 11:08:25 +01:00
Takashi Iwai 888ea7d5ac ALSA: usb-audio: Fix races at disconnection in mixer_quirks.c
Similar like the previous commit, cover with chip->shutdown_rwsem
and chip->shutdown checks.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:05 +01:00
Takashi Iwai 34f3c89fda ALSA: usb-audio: Use rwsem for disconnect protection
Replace mutex with rwsem for codec->shutdown protection so that
concurrent accesses are allowed.

Also add the protection to snd_usb_autosuspend() and
snd_usb_autoresume(), too.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:07:00 +01:00
Takashi Iwai 978520b75f ALSA: usb-audio: Fix races at disconnection
Close some races at disconnection of a USB audio device by adding the
chip->shutdown_mutex and chip->shutdown check at appropriate places.

The spots to put bandaids are:
- PCM prepare, hw_params and hw_free
- where the usb device is accessed for communication or get speed, in
 mixer.c and others; the device speed is now cached in subs->speed
 instead of accessing to chip->dev

The accesses in PCM open and close don't need the mutex protection
because these are already handled in the core PCM disconnection code.

The autosuspend/autoresume codes are still uncovered by this patch
because of possible mutex deadlocks.  They'll be covered by the
upcoming change to rwsem.

Also the mixer codes are untouched, too.  These will be fixed in
another patch, too.

Reported-by: Matthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-30 11:06:54 +01:00
Kees Cook f598158aa4 ALSA: sound/usb: remove CONFIG_EXPERIMENTAL
This config item has not carried much meaning for a while now and is
almost always enabled by default. As agreed during the Linux kernel
summit, remove it.

Signed-off-by: Kees Cook <keescook@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-25 00:17:47 +02:00
Didier Villevalois c902466800 ALSA: usb-audio: Add quirk for Reloop Play
The Reloop Audio needs a fixed endpoint quirk with S24_3LE format and
UAC_EP_CS_ATTR_SAMPLE_RATE attribute.

Signed-off-by: Didier Villevalois <ptitjes@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-23 16:38:14 +02:00
Pete Leigh 7a75e742fa ALSA: usb-audio: USB audio quirk for Roland VG-99 advanced mode
Without this quirk the VG-99 will work in standard mode (set under
USB on System menu page 2) giving 16 bits at 44.1 Khz audio in/out
but no midi, and is not recognised when set to advanced mode.

After applying this, I can also use the VG-99 in advanced mode: 24
24 bits audio in/out at 44.1 Khz, and midi in/out. Sysex is so far
untested.

In standard mode, the device appears with ID 0x00b3, so the
behaviour isn't affected by this quirk.

Thanks to Clemens Ladisch for simplifying and correcting my initial
attempt!

Signed-off-by: Pete Leigh <pete.leigh@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-21 12:05:03 +02:00
Wei Yongjun 950f40fdd4 ALSA: snd-usb: remove unused variable in init_pitch_v2()
The variable ep is initialized but never used
otherwise, so remove the unused variable.

dpatch engine is used to auto generate this patch.
(https://github.com/weiyj/dpatch)

Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-21 10:43:27 +02:00
Linus Torvalds 2fc07efa22 Sound updates #2 for 3.7-rc1
This update contains a few cleanup works, regression/stable fixes
 gathered since the last pull request.
 
 - Clean up with generic hd-audio jack handling code by David
   Henningsson
 - A few regression fixes for standardized HD-audio auto-parser
 - Misc clean-up and small fixes
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates #2 from Takashi Iwai:
 "This update contains a few cleanup works, regression/stable fixes
  gathered since the last pull request.

   - Clean up with generic hd-audio jack handling code by David
     Henningsson
   - A few regression fixes for standardized HD-audio auto-parser
   - Misc clean-up and small fixes"

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - do not detect jack on internal speakers for Realtek
  ALSA: hda - Fix missing beep on ASUS X43U notebook
  ALSA: hda - Remove AZX_DCAPS_POSFIX_COMBO
  ALSA: hda - Warn an allocation for an uninitialized array
  ALSA: hda/cirrus - Add missing init/free of hda_gen_spec
  ALSA: hda - Fix memory leaks at error path in patch_cirrus.c
  ALSA: hda - Add missing hda_gen_spec to struct via_spec
  ALSA: hda - remove "Mic Jack Mode" for headset jacks (Latitude Exx30)
  ALSA: hda - make Cirrus codec use generic unsol event handler
  ALSA: hda - make VIA codec use generic unsol event handler
  ALSA: hda - Remove dead GPIO code for VIA codec
  ALSA: usb-audio: Add TASCAM US122 MKII playback
2012-10-12 12:31:28 +09:00
Konstantin Khlebnikov 314e51b985 mm: kill vma flag VM_RESERVED and mm->reserved_vm counter
A long time ago, in v2.4, VM_RESERVED kept swapout process off VMA,
currently it lost original meaning but still has some effects:

 | effect                 | alternative flags
-+------------------------+---------------------------------------------
1| account as reserved_vm | VM_IO
2| skip in core dump      | VM_IO, VM_DONTDUMP
3| do not merge or expand | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP
4| do not mlock           | VM_IO, VM_DONTEXPAND, VM_HUGETLB, VM_PFNMAP

This patch removes reserved_vm counter from mm_struct.  Seems like nobody
cares about it, it does not exported into userspace directly, it only
reduces total_vm showed in proc.

Thus VM_RESERVED can be replaced with VM_IO or pair VM_DONTEXPAND | VM_DONTDUMP.

remap_pfn_range() and io_remap_pfn_range() set VM_IO|VM_DONTEXPAND|VM_DONTDUMP.
remap_vmalloc_range() set VM_DONTEXPAND | VM_DONTDUMP.

[akpm@linux-foundation.org: drivers/vfio/pci/vfio_pci.c fixup]
Signed-off-by: Konstantin Khlebnikov <khlebnikov@openvz.org>
Cc: Alexander Viro <viro@zeniv.linux.org.uk>
Cc: Carsten Otte <cotte@de.ibm.com>
Cc: Chris Metcalf <cmetcalf@tilera.com>
Cc: Cyrill Gorcunov <gorcunov@openvz.org>
Cc: Eric Paris <eparis@redhat.com>
Cc: H. Peter Anvin <hpa@zytor.com>
Cc: Hugh Dickins <hughd@google.com>
Cc: Ingo Molnar <mingo@redhat.com>
Cc: James Morris <james.l.morris@oracle.com>
Cc: Jason Baron <jbaron@redhat.com>
Cc: Kentaro Takeda <takedakn@nttdata.co.jp>
Cc: Matt Helsley <matthltc@us.ibm.com>
Cc: Nick Piggin <npiggin@kernel.dk>
Cc: Oleg Nesterov <oleg@redhat.com>
Cc: Peter Zijlstra <a.p.zijlstra@chello.nl>
Cc: Robert Richter <robert.richter@amd.com>
Cc: Suresh Siddha <suresh.b.siddha@intel.com>
Cc: Tetsuo Handa <penguin-kernel@I-love.SAKURA.ne.jp>
Cc: Venkatesh Pallipadi <venki@google.com>
Acked-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2012-10-09 16:22:19 +09:00
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
Oto Petřík 613769fcab ALSA: usb-audio: Add TASCAM US122 MKII playback
Added quirk to provide at least playback-only support.

Signed-off-by: Oto Petrik <oto.petrik@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-08 15:16:33 +02:00
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
David Henningsson c10514394e ALSA: usb - disable broken hw volume for Tenx TP6911
While going through Ubuntu bugs, I discovered this patch being
posted and a confirmation that the patch works as expected.

Finding out how the hw volume really works would be preferrable
to just disabling the broken one, but this would be better than
nothing.

Credit: sndfnsdfin (qawsnews)
BugLink: https://bugs.launchpad.net/bugs/559939
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-20 10:48:47 +02:00
Takashi Iwai 384dc085c3 ALSA: usb-audio: Avoid unnecessary EP setups in prepare
The recent fix for USB suspend breakage moved the code to set up EP
from hw_params to prepare, but it means also the EP setup might be
called multiple times unnecessarily because the prepare callback can
be called multiple times without starting the stream (e.g. OSS
emulation).

This patch adds a new flag to struct snd_usb_substream indicating
whether the setup of EP is required, and do it only when necessary,
i.e. right after hw_params or suspend.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:08:16 +02:00
Dylan Reid 61a709504b ALSA: usb-audio: Move configuration to prepare.
Move interface and endpoint configuration from hw_params to prepare
callback.  During system suspend/resume when the USB device power isn't
cycled the interface and endpoint configuration need to be set before
audio playback can continue.  Resume involves another call to prepare
but not to hw_params, moving it here allows a playing stream to continue
after resume.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:08:11 +02:00
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
Dylan Reid 715a170563 ALSA: usb-audio: set period_bytes in substream.
Set the peiod_bytes member of snd_usb_substream.  It was no longer being
set, but will be needed to resume properly in a future commit.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-19 08:07:34 +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
Takashi Iwai 1213a205f9 ALSA: usb-audio: Fix bogus error messages for delay accounting
The recent fix for the missing fine delayed time adjustment gives
strange error messages at each start of the playback stream, such as
  delay: estimated 0, actual 352
  delay: estimated 353, actual 705

These come from the sanity check in retire_playback_urb().  Before the
stream is activated via start_endpoints(), a few silent packets have
been already sent.  And at this point the delay account is still in
the state as if the new packets are just queued, so the driver gets
confused and spews the bogus error messages.

For fixing the issue, we just need to check whether the received
packet is valid, whether it's zero sized or not.

Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-06 15:00:15 +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
Marko Friedemann c05fce586d ALSA: USB: Support for (original) Xbox Communicator
Added support for Xbox Communicator to USB quirks.

Signed-off-by: Marko Friedemann <mfr@bmx-chemnitz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-09-03 10:14:25 +02:00
Daniel Mack 2e4a263ca8 ALSA: snd-usb: fix cross-interface streaming devices
Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface")
saved us some unnecessary calls to snd_usb_set_interface() but ignored
the fact that there is at least one device out there which operates on
two endpoint in different interfaces simultaniously.

Take care for this by catching the case where data and sync endpoints
are located on different interfaces and calling snd_usb_set_interface()
between the start of the two endpoints.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Robert M. Albrecht <linux@romal.de>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:04:53 +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 fbcfbf5f67 ALSA: snd-usb: restore delay information
Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB
frame counter") were unfortunately lost during the refactoring of the
snd-usb driver in 3.5.

This patch adds them back, restoring the correct delay information
behaviour.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 21:03:08 +02:00
Pavel Roskin 03d2f44e96 ALSA: snd-usb: use list_for_each_safe for endpoint resources
snd_usb_endpoint_free() frees the structure that contains its argument.

Signed-off-by: Pavel Roskin <proski@gnu.org>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-31 18:17:45 +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
Takashi Iwai 48ee7cb8b4 ALSA: usb-audio: Remove obsoleted fields in struct snd_usb_substream
The two entries are duplicated in struct snd_usb_endpoint.
Seems forgotten in the last clean-up.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-28 16:30:02 -07:00
Takashi Iwai ddf83485d7 Merge branch 'for-linus' into for-next
Conflicts:
	sound/pci/hda/hda_codec.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-20 22:14:26 +02:00
Takashi Iwai e9ba389c5f ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream
A PCM capture stream on usb-audio causes a scheduling-while-atomic
BUG, as reported in the bugzilla entry below.  It's because
snd_usb_endpoint_start() is called at first at trigger START for a
capture stream, and this function contains the left-over EP
deactivation codes.  The problem doesn't happen for a playback stream
because the function is called at PCM prepare time, which can sleep.

This patch fixes the BUG by moving the EP deactivation code into the
PCM prepare callback.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-16 08:04:07 +02:00
Andy Shevchenko 793ea49c47 ALSA: print small buffers via %*ph[C]
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-06 11:09:50 +02:00
Daniel Mack aff252a848 ALSA: snd-usb: fix clock source validity index
uac_clock_source_is_valid() uses the control selector value to access
the bmControls bitmap of the clock source unit. This is wrong, as
control selector values start from 1, while the bitmap uses all
available bits.

In other words, "Clock Validity Control" is stored in D3..2, not D5..4
of the clock selector unit's bmControls.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Andreas Koch <andreas@akdesigninc.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-08-01 10:24:16 +02:00
Takashi Iwai f0913cd16e Merge branch 'topic/misc' into for-next
Generic updates for sound 3.6
2012-07-18 13:53:29 +02:00
Daniel Mack 68e67f40b7 ALSA: snd-usb: move calls to usb_set_interface
The rework of the snd-usb endpoint logic moved the calls to
snd_usb_set_interface() into the snd_usb_endpoint implemenation. This
changed the order in which these calls are issued to the device, and
thereby caused regressions for some webcams.

Fix this by moving the calls back to pcm.c for now to make it work again
and use snd_usb_endpoint_activate() to really tear down all remaining
URBs in the flight, consequently fixing another regression caused by USB
packets on the wire after altsetting 0 has been selected.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Philipp Dreimann <philipp@dreimann.net>
Reported-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-13 09:31:42 +02:00
Takashi Iwai 9e9b594661 ALSA: usb-audio: Fix the first PCM interface assignment
In the new PCM streaming logic, the interface number is assigned to
usb stream instance (subs->interface) after the format and rate setups
are succeeded, but some codes are still passing subs->interface as the
reference to helper functions.  This leads to initializing with an
invalid iface number (-1).

This patch replaces the wrong references with the ones from the target
fmt correctly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-07-06 08:11:43 +02:00
Daniel Mack da185443c1 ALSA: snd-usb-caiaq: initialize card pointer
Fixes the following warning:

  CC [M]  sound/usb/caiaq/device.o
sound/usb/caiaq/device.c: In function ‘snd_probe’:
sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used
uninitialized in this function [-Wmaybe-uninitialized]

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-27 12:26:19 +02:00
Clemens Ladisch 74953e2010 ALSA: usb-audio: add BOSS GT-100 support
Reported-by: John McFarland <mcfarljm@gmail.com>
Tested-by: John McFarland <mcfarljm@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-25 11:11:24 +02:00
Oleksij Rempel b64a1ba9d3 ALSA: snd_usb_audio: ignore ctrl errors on QuickCam Pro for Notebooks
This webcam works mostly ok, exept with skype.
Skype sends lots of ctrl messages to dynamically ajust
record level. If for some reasons it pokes some error
every thing goes broken:
- first pulseaudio blocks sound for all apps
- then video is reseted
- then skype freez

dmesg has lots of messages like:
cannot set freq 16000 to ep 0x86"

Setting ignore_ctl_error=1 fixes this problem.

Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-20 08:29:08 +02:00
Oleksij Rempel 05b9afd5b7 ALSA: snd_usb_audio: ignore ctrl errors on QuickCam E3500
if this cam is pluged in, pulse audio can't initiate capture
device.
dmesg has lots of messages like:
"cannot set freq 16000 to ep 0x86"

Setting ignore_ctl_error=1 fixes this problem.

Signed-off-by: Oleksij Rempel <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-20 08:28:57 +02:00
Daniel Mack 0b1d8e0908 ALSA: 6fire: use NULL instead of 0 for pointer assignment
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-18 09:36:38 +02:00
Daniel Mack afe25967ec ALSA: snd-usb: make snd_usb_substream_capture_trigger static
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-18 09:32:53 +02:00
Daniel Mack 7fb75db139 ALSA: snd-usb: fix sync pipe check
Fix a bogus sanity check for sync pipe in pcm.c. This flaw was
introduced during the streaming logic refactorization.

While at it, improve the error messages that are generated in such cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: <ben@b1c1l1.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-18 08:36:36 +02:00
Mark Hills 989b01385f ALSA: usb-audio: Convert table to preferred C99 format
Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-11 12:49:56 +02:00
Mark Hills b71dad181a ALSA: usb-audio: Use a table of mixer controls
Allow mixer controls to be provided clearly in a table, to avoid
quantity of error checking at each use.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-11 12:49:43 +02:00
Takashi Iwai 8260ef075b ALSA: usb-audio: Fix substream assignments
In 3.5 kernel, the endpoint is assigned dynamically for the
substreams, but the PCM assignment still checks the presence of the
endpoint pointer.  This ended up in duplicated PCM substream creations
at probing time, resulting in kernel warnings like:

WARNING: at fs/proc/generic.c:586 proc_register+0x169/0x1a6()
Pid: 1152, comm: modprobe Not tainted 3.5.0-rc1-00110-g71fae7e #2
Call Trace:
 [<ffffffff8102a400>] warn_slowpath_common+0x83/0x9c
 [<ffffffff8102a4bc>] warn_slowpath_fmt+0x46/0x48
 [<ffffffff813829ad>] ? add_preempt_count+0x39/0x3b
 [<ffffffff811292f0>] proc_register+0x169/0x1a6
 [<ffffffff8112962e>] create_proc_entry+0x74/0x8c
 [<ffffffffa018eb63>] snd_info_register+0x3e/0xc3 [snd]
 [<ffffffffa01fde2e>] snd_pcm_new_stream+0xb1/0x404 [snd_pcm]
 [<ffffffffa024861f>] snd_usb_add_audio_stream+0xd2/0x230 [snd_usb_audio]
 [<ffffffffa0241d33>] ? snd_usb_parse_audio_format+0x252/0x34f [snd_usb_audio]
 [<ffffffff810d6b17>] ? kmem_cache_alloc_trace+0xab/0xbb
 [<ffffffffa0248c29>] snd_usb_parse_audio_interface+0x4ac/0x567 [snd_usb_audio]
 [<ffffffffa023f0ff>] snd_usb_create_stream+0xe9/0x125 [snd_usb_audio]
 [<ffffffffa023f9b1>] usb_audio_probe+0x62a/0x72c [snd_usb_audio]
 .....

This patch fixes the regression by checking the fixed endpoint number
for each substream instead of the endpoint pointer.

Reported-and-tested-by: Jamie Heilman <jamie@audible.transient.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-06-08 09:01:37 +02:00
Clemens Ladisch 5cd5d7c449 ALSA: usb-audio: fix rate_list memory leak
The array of sample rates is reallocated every time when opening
the PCM device, but was freed only once when unplugging the device.

Reported-by: "Alexander E. Patrakov" <patrakov@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-31 10:25:44 +02:00
Daniel Mack 97f8d3b650 ALSA: snd-usb: fix stream info output in /proc
Set some substream struct members to make the proc interface code work
again.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-21 12:51:08 +02:00
Takashi Iwai e182534d4b ALSA: usb-audio - Call get_min_max_*() after determining the name string
get_min_max_with_quirks() must be called after the control id name
string is determined, but the current code changes the id name string
after calling the function.

Reported-by: Christian Melki <christian.melki@ericsson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-15 08:35:00 +02:00
Mark Hills 7df4a691fb ALSA: usb-audio: Fix comment
Explained by Takashi in <s5hfwbtmz0q.wl%tiwai@suse.de>

> The reason is because get_min_max*() isn't called in the place you
> created these controls, and get_min_max() would be called only for
> integer volumes later even if uninitialized.  A short cut for booleans.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-05-11 21:27:36 +02:00
Daniel Mack 07a5e9d4fd ALSA: snd-usb: fix some typos in endpoint.c documentation
Also be more specific about some details while at it.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 20:16:18 +02:00
Andrew Morton 68853fa30c ALSA: usb-audio: sound/usb/endpoint.c: suppress warning
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function

Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:10:10 +02:00
Takashi Iwai baba2e0d2b ALSA: usb-audio: Add missing error checks in snd_ebox44_create_mixer()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:07:38 +02:00
Felix Homann d34bf14851 ALSA: usb-audio: M-Audio Fast Track Ultra: Add effect controls
This adds controls for the effects section on the FTU devices.
Some of these controls need volume quirks. They are added to
mixer.c.

[fixed missing break by tiwai]

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:06:06 +02:00
Felix Homann cfe8f97c82 ALSA: usb-audio: Rename Fast Track Ultra mixer quirk functions
This is in preparation for more FTU controls to come.
Should help keeping names a bit shorter.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:02:11 +02:00
Felix Homann 25ee7ef8fa ALSA: usb-audio: Add TLV to M-Audio Fast Track Ultra controls
This adds db gain information to M-Audio Fast Track Ultra (8R) devices.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:01:46 +02:00
Felix Homann 285de9c08b ALSA: usb-audio: Rename and export mixer_vol_tlv
Rename mixer_vol_tlv to snd_usb_mixer_vol_tlv and export it to make
it reuseable in mixer_quirks.c.

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:01:27 +02:00
Felix Homann 8a4d1d397b ALSA: usb-audio: Unify M-Audio Fast Track Ultra and Ebox-44 mixer quirks.
Merge snd_maudio_ftu_create_ctl() and snd_ebox44_create_ctl() into
snd_create_std_mono_ctl().
As opposed to the ftu and ebox-44 specific functions, a TLV callback
can be specified for controls created by snd_create_std_mono_ctl().

[fixed minor checkpatch.pl warnings by tiwai]

Signed-off-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-24 08:00:45 +02:00
Daniel Mack c89a5d9cac ALSA: snd-usb: remove refactorization left-overs
Drop some struct members and definitions that became obsolete during
the refactorization of the driver.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-21 17:40:28 +02:00
Takashi Iwai 56599bb020 Merge branch 'topic/usb-endpoint' into topic/misc 2012-04-18 07:57:32 +02:00
Mark Hills 7536c301f8 ALSA: snd-usb-audio: Replace mixer for Electrix Ebox-44
The mixer units from the firmware are corrupt, and even where they
are valid they presents mono controls as L and R channels of
stereo.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-15 15:40:08 +02:00
Mark Hills 284a8dd6f0 ALSA: snd-usb-audio: Skip un-parseable mixer units instead of erroring
Some interfaces reference endpoints which do not exists. To
accomodate these, do not fail completely, but skip over them.

This allows the Electrix Ebox-44 with earlier firmware to be
detected and used for audio.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-15 15:39:55 +02:00
Takashi Iwai 22026c1a7b ALSA: usb: Remove obsoleted fields from struct snd_usb_substream
Many fields have been moved to struct snd_usb_endpoint.
Also fix the proc output to correspond to the new structure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 12:57:39 +02:00
Takashi Iwai 85f71932e5 ALSA: usb: Fix fill_max flag set
ep->fill_max is a 1 bit flag, thus it has to be boolean.
  sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
  sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 12:41:54 +02:00
Takashi Iwai c5ee4ec828 ALSA: usb: Remove unused variable
sound/usb/endpoint.c: In function ‘deactivate_urbs’:
sound/usb/endpoint.c:520:16: warning: unused variable ‘flags’ [-Wunused-variable]

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:27:28 +02:00
Daniel Mack 94c27215bc ALSA: snd-usb: add some documentation
Document the new streaming code and some of the functions so that
contributers can catch up easier.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:25:24 +02:00
Daniel Mack c75a8a7ae5 ALSA: snd-usb: add support for implicit feedback
Implicit feedback is a streaming mode that does not rely on dedicated
sync endpoints but uses the information provided by record streams to
clock output streams. Now that the streaming logic is decoupled from the
PCM streams, this is easy to implement.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:32 +02:00
Daniel Mack d399ff9593 ALSA: snd-usb: remove old streaming logic
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:23 +02:00
Daniel Mack edcd3633e7 ALSA: snd-usb: switch over to new endpoint streaming logic
With the previous commit that added the new streaming model, all
endpoint and streaming related code is now in endpoint.c, and pcm.c
only acts as a wrapper for handling the packet's payload.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:24:08 +02:00
Daniel Mack 8fdff6a319 ALSA: snd-usb: implement new endpoint streaming model
This patch adds a new generic streaming logic for audio over USB.

It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.

A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.

With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.

In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:23:42 +02:00
Daniel Mack 596580d0ee ALSA: snd-usb: add snd_usb_audio-wide mutex
This is needed for new card-wide list operations.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-04-13 10:21:55 +02:00
Takashi Iwai 44c76a960a Merge branch 'topic/misc' into for-linus 2012-03-18 18:22:33 +01:00
Takashi Iwai 0717d0f5d2 ALSA: usb-audio - Fix build error by consitification of rate list
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-03-15 16:14:38 +01:00
Torsten Schenk adef39c0ea ALSA: snd-usb-6fire: Select missing SND_VMASTER option in Kconfig
Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-25 11:07:19 +01:00
Torsten Schenk 06bb4e7435 ALSA: snd-usb-6fire: add analog input volume control
Add a stereo volume control for analog input channel pair 1/2.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:51:26 +01:00
Torsten Schenk d97c735a10 ALSA: snd-usb-6fire: add mute control for analog outputs
Add a mute control for every analog output channel.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:51:16 +01:00
Torsten Schenk f90ffbf3c6 ALSA: snd-usb-6fire: add individual volume control for analog channels
Add a stereo volume control for every analog output pair 1/2, 3/4, 5/6.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:51:06 +01:00
Torsten Schenk 8e247a9c90 ALSA: snd-usb-6fire: add tlv to controls
Remove the soft log-conversion and add a dB scale according to
the DAC documentation instead.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:50:56 +01:00
Torsten Schenk c596758f57 ALSA: snd-usb-6fire: remove driver version information
Remove unused driver version information from the individual files.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 15:50:45 +01:00
Mark Hills cb74eb15ac ALSA: snd-usb-caiaq: Fix the return of XRUN
Commit 3702b08 added a lock, but did not account for the case of
SNDRV_PCM_POS_XRUN, which would get immediately overwritten.

This could be bundled into one if-else-if statement, but the goto
helps to clarify the 'exceptional' case.

Thanks to Andreas Pape for spotting this.

Signed-off-by: Mark Hills <mark@pogo.org.uk>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-22 08:34:58 +01:00
Xi Wang 8866f405ef ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk()
A malicious USB device could feed in a large nr_rates value.  This would
cause the subsequent call to kmemdup() to allocate a smaller buffer than
expected, leading to out-of-bounds access.

This patch validates the nr_rates value and reuses the limit introduced
in commit 4fa0e81b ("ALSA: usb-audio: fix possible hang and overflow
in parse_uac2_sample_rate_range()").

Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-15 14:58:15 +01:00
Masanari Iida 6e8d5d2f17 ALSA: usx2y: Fix typo in usbusx2yaudio.c and usx2yhwdeppcm.c
Correct spelling "propably" to "probably" and "activ" to "active"
in sound/usb/usx2y/usbusx2yaudio.c and usx2yhwdeppcm.c

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-15 14:56:11 +01:00
Clemens Ladisch 927c9423dd ALSA: usb-audio: add Edirol UM-3G support
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-02-08 09:46:34 +01:00
Linus Torvalds a429638cac Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
  ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
  ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
  ALSA: usb-audio: add Yamaha MOX6/MOX8 support
  ALSA: virtuoso: add S/PDIF input support for all Xonars
  ALSA: ice1724 - Support for ooAoo SQ210a
  ALSA: ice1724 - Allow card info based on model only
  ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
  ALSA: hdspm - Provide unique driver id based on card serial
  ASoC: Dynamically allocate the rtd device for a non-empty release()
  ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
  ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
  ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
  ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
  ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
  ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
  ALSA: hda/cirrus - support for iMac12,2 model
  ASoC: cx20442: add bias control over a platform provided regulator
  ALSA: usb-audio - Avoid flood of frame-active debug messages
  ALSA: snd-usb-us122l: Delete calls to preempt_disable
  mfd: Put WM8994 into cache only mode when suspending
  ...

Fix up trivial conflicts in:
 - arch/arm/mach-s3c64xx/mach-crag6410.c:
	renamed speyside_wm8962 to tobermory, added littlemill right
	next to it
 - drivers/base/regmap/{regcache.c,regmap.c}:
	duplicate diff that had already come in with other changes in
	the regmap tree
2012-01-12 08:00:30 -08:00
Clemens Ladisch 8c3f5d8a9b ALSA: usb-audio: add Yamaha MOX6/MOX8 support
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-11 09:24:53 +01:00
Takashi Iwai 80c8a2a372 ALSA: usb-audio - Avoid flood of frame-active debug messages
With some buggy devices, the usb-audio driver may give "frame xxx active"
kernel messages too often.  Better to keep it as debug-only using
snd_printdd(), and also add the rate-limit for avoiding floods.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738681

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 11:40:46 +01:00
Karsten Wiese d0f3a2eb90 ALSA: snd-usb-us122l: Delete calls to preempt_disable
They are not needed here.

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-09 11:31:30 +01:00
Xi Wang 4fa0e81b83 ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range()
A malicious USB device may feed in carefully crafted min/max/res values,
so that the inner loop in parse_uac2_sample_rate_range() could run for
a long time or even never terminate, e.g., given max = INT_MAX.

Also nr_rates could be a large integer, which causes an integer overflow
in the subsequent call to kmalloc() in parse_audio_format_rates_v2().
Thus, kmalloc() would allocate a smaller buffer than expected, leading
to a memory corruption.

To exploit the two vulnerabilities, an attacker needs physical access
to the machine to plug in a malicious USB device.

This patch makes two changes.

1) The type of "rate" is changed to unsigned int, so that the loop could
   stop once "rate" is larger than INT_MAX.

2) Limit nr_rates to 1024.

Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-01-08 16:03:12 +01:00
Greg Kroah-Hartman ff4b8a57f0 Merge branch 'driver-core-next' into Linux 3.2
This resolves the conflict in the arch/arm/mach-s3c64xx/s3c6400.c file,
and it fixes the build error in the arch/x86/kernel/microcode_core.c
file, that the merge did not catch.

The microcode_core.c patch was provided by Stephen Rothwell
<sfr@canb.auug.org.au> who was invaluable in the merge issues involved
with the large sysdev removal process in the driver-core tree.

Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2012-01-06 11:42:52 -08:00
Rusty Russell a67ff6a540 ALSA: module_param: make bool parameters really bool
module_param(bool) used to counter-intuitively take an int.  In
fddd5201 (mid-2009) we allowed bool or int/unsigned int using a messy
trick.

It's time to remove the int/unsigned int option.  For this version
it'll simply give a warning, but it'll break next kernel version.

Signed-off-by: Rusty Russell <rusty@rustcorp.com.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-19 10:34:41 +01:00
Sergiusz Urbaniak 1bba160a07 ALSA: snd-usb: added VOX ToneLab ST midi handling
Signed-off-by: Sergiusz Urbaniak <sergiusz.urbaniak@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-12-12 12:49:02 +01:00
John F Leach ae7cc709f2 ALSA: usb-audio - Support for Roland GAIA SH-01 Synthesizer
Added table quirks entry for Roland GAIA SH-01 Synthesizer based upon
Roland SH-201 table entry as template.  USB MIDI and audio was tested
with Muse and Audacity.

Signed-off-by: John F Leach <jfleach@jfleach.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-29 08:23:15 +01:00
Greg Kroah-Hartman 424f0750ed USB: convert sound/* to use module_usb_driver()
This converts the drivers in sound/* to use the
module_usb_driver() macro which makes the code smaller and a bit
simpler.

Added bonus is that it removes some unneeded kernel log messages about
drivers loading and/or unloading.

Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Daniel Mack <zonque@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Cc: Torsten Schenk <torsten.schenk@zoho.com>
Cc: Paul Gortmaker <paul.gortmaker@windriver.com>
Cc: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-18 09:50:44 -08:00
Thomas Meyer 43df2a57b7 ALSA: usb-audio: Use kmemdup rather than duplicating its implementation
Use kmemdup rather than duplicating its implementation

The semantic patch that makes this change is available
in scripts/coccinelle/api/memdup.cocci.

Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-10 19:51:45 +01:00
Alexey Fisher 55c0008be6 ALSA: snd_usb_audio: add Logitech HD Webcam c510 to quirk-384
Logitech HD Webcam c510 provide wrong mixer resolution.
Add it to "res = 384" quirk.

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-09 12:22:38 +01:00
Takashi Iwai dcaaf9f2c1 ALSA: usb-audio - Fix the missing volume quirks at delayed init
In the recent usb-audio driver, the initialization of volume ranges
may be delayed when the device doesn't respond well at the probing time.
But the volume quirks for certain devices are applied only in
mixer_ctl_feature_info() thus only at the very first probe and will be
missing when the volume range is initialized later.

This patch moves the volume quirk code to be always called from the
volume-range extraction (get_min_max()), so that the quirks are properly
applied in the later init time.

Reported-and-tested-by: Alexey Fisher <bug-track@fisher-privat.net>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-08 17:50:27 +01:00
Linus Torvalds 32aaeffbd4 Merge branch 'modsplit-Oct31_2011' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux
* 'modsplit-Oct31_2011' of git://git.kernel.org/pub/scm/linux/kernel/git/paulg/linux: (230 commits)
  Revert "tracing: Include module.h in define_trace.h"
  irq: don't put module.h into irq.h for tracking irqgen modules.
  bluetooth: macroize two small inlines to avoid module.h
  ip_vs.h: fix implicit use of module_get/module_put from module.h
  nf_conntrack.h: fix up fallout from implicit moduleparam.h presence
  include: replace linux/module.h with "struct module" wherever possible
  include: convert various register fcns to macros to avoid include chaining
  crypto.h: remove unused crypto_tfm_alg_modname() inline
  uwb.h: fix implicit use of asm/page.h for PAGE_SIZE
  pm_runtime.h: explicitly requires notifier.h
  linux/dmaengine.h: fix implicit use of bitmap.h and asm/page.h
  miscdevice.h: fix up implicit use of lists and types
  stop_machine.h: fix implicit use of smp.h for smp_processor_id
  of: fix implicit use of errno.h in include/linux/of.h
  of_platform.h: delete needless include <linux/module.h>
  acpi: remove module.h include from platform/aclinux.h
  miscdevice.h: delete unnecessary inclusion of module.h
  device_cgroup.h: delete needless include <linux/module.h>
  net: sch_generic remove redundant use of <linux/module.h>
  net: inet_timewait_sock doesnt need <linux/module.h>
  ...

Fix up trivial conflicts (other header files, and  removal of the ab3550 mfd driver) in
 - drivers/media/dvb/frontends/dibx000_common.c
 - drivers/media/video/{mt9m111.c,ov6650.c}
 - drivers/mfd/ab3550-core.c
 - include/linux/dmaengine.h
2011-11-06 19:44:47 -08:00
Linus Torvalds 9991357259 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Revert the check of NO_PRESENCE pincfg default bit
  ALSA: hda - Fix a regression for DMA-position check with CA0110
  ALSA: hda - Fix silent output regression with ALC861
  ALSA: control: remove compilation warning on 32-bit
  ALSA: ua101: fix crash when unplugging
2011-11-06 12:14:22 -08:00
Clemens Ladisch 862a6244eb ALSA: ua101: fix crash when unplugging
If the device is unplugged while running, it is possible for a PCM
device to be closed after the disconnect callback has returned.  This
means that kill_stream_urb() and disable_iso_interface() would try to
access already-invalid or freed USB data structures.

The function free_usb_related_resources() was intended to prevent this,
but forgot to clear the affected variables.

Reported-and-tested-by: Olivier Courtay <olivier@courtay.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: 2.6.33+ <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-11-06 11:21:42 +01:00
Paul Gortmaker da155d5b40 sound: Add module.h to the previously silent sound users
Lots of sound drivers were getting module.h via the implicit presence
of it in <linux/device.h> but we are going to clean that up.  So
fix up those users now.

Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
2011-10-31 19:31:21 -04:00
Paul Gortmaker 31623caaf0 sound: add moduleparam.h to users of module_param/MODULE_PARM_DESC
These files were getting access to these two via the implicit
presence of moduleparam.h everywhere.  But that is being fixed, so
get these guys what they need in advance.

Signed-off-by: Paul Gortmaker <paul.gortmaker@windriver.com>
2011-10-31 19:31:20 -04:00
Linus Torvalds 68d99b2c8e Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
  ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
  ALSA: hda - Keep EAPD turned on for old Conexant chips
  ALSA: hda/realtek - Fix missing volume controls with ALC260
  ASoC: wm8940: Properly set codec->dapm.bias_level
  ALSA: hda - Fix pin-config for ASUS W90V
  ALSA: hda - Fix surround/CLFE headphone and speaker pins order
  ALSA: hda - Fix typo
  ALSA: Update the sound git tree URL
  ALSA: HDA: Add new revision for ALC662
  ASoC: max98095: Convert codec->hw_write to snd_soc_write
  ASoC: keep pointer to resource so it can be freed
  ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
  ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
  ASoC: da7210: Add support for line out and DAC
  ASoC: da7210: Add support for DAPM
  ALSA: hda/realtek - Fix DAC assignments of multiple speakers
  ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
  ASoC: Set sgtl5000->ldo in ldo_regulator_register
  ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
  ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
  ...
2011-10-28 14:25:01 -07:00
Takashi Iwai d226657022 Merge branch 'topic/misc' into for-linus 2011-10-26 23:51:43 +02:00
Takashi Iwai d09c06c6fc ALSA: usb-audio - Fix possible access over audio_feature_info[] array
The audio_feature_info[] array should contain all entries for UAC2_FU_*,
but currently a few last entries are missing.  Even though, the driver
tries to probe these entries in parse_audio_feature_unit() and may
access the range over the array.  This patch fixes the bug by limiting
the loop size properly using ARRAY_SIZE() instead of a hard-coded
magic number.

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-13 08:19:09 +02:00
William Light e653510a27 ALSA: snd-usb-caiaq: Add support for Maschine
This adds partial support for the Maschine controller by Native Instruments.
Supported now are the 1x1 MIDI interface and the 41 buttons, 11 endless
rotary encoders, and 16 pressure-sensitive drum pads. Still to work on are the
dimmable LEDs and the two monochrome screens.

Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-13 08:16:46 +02:00
William Light 3d37fbe441 ALSA: snd-usb-caiaq: Fix NULL dereference in input.c
There was a case where a newly-registered input device could be opened before
a necessary variable in the device structure was set. When code tried to use
the variable in the URB reply callback, it would cause an Oops.

This fix sets the aforementioned variable before calling input_register_device.

Signed-off-by: William Light <wrl@illest.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-10-13 08:16:42 +02:00
Rafael J. Wysocki d727b60659 Merge branch 'pm-runtime' into pm-for-linus
* pm-runtime:
  PM / Tracing: build rpm-traces.c only if CONFIG_PM_RUNTIME is set
  PM / Runtime: Replace dev_dbg() with trace_rpm_*()
  PM / Runtime: Introduce trace points for tracing rpm_* functions
  PM / Runtime: Don't run callbacks under lock for power.irq_safe set
  USB: Add wakeup info to debugging messages
  PM / Runtime: pm_runtime_idle() can be called in atomic context
  PM / Runtime: Add macro to test for runtime PM events
  PM / Runtime: Add might_sleep() to runtime PM functions
2011-10-07 23:16:55 +02:00
Clemens Ladisch 17d900c4a1 ALSA: usb-audio: increase control transfer timeout
There are certain devices that are reportedly so slow that they need
more than 100 ms to handle control transfers.  Therefore, increase the
timeout in mixer(_quirks).c to 1000 ms.

The timeout parameter of snd_usb_ctl_msg() is now constant, so we can
drop it.

Reported-by: Felipe Balbi <balbi@ti.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-27 09:21:48 +02:00
Thomas Pfaff 61a6a108d1 ALSA: usb-audio: Check for possible chip NULL pointer before clearing probing flag
Before clearing the probing flag in the error exit path, check that the
chip pointer is not NULL.

Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-26 15:48:47 +02:00
Thomas Pfaff 362e4e49ab ALSA: usb-audio - clear chip->probing on error exit
The Terratec Aureon 5.1 USB sound card support is broken since kernel
2.6.39.
2.6.39 introduced power management support for USB sound cards that added
a probing flag in struct snd_usb_audio.

During the probe of the card it gives following error message :

usb 7-2: new full speed USB device number 2 using uhci_hcd
cannot find UAC_HEADER
snd-usb-audio: probe of 7-2:1.3 failed with error -5
input: USB Audio as
/devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.3/input/input6
generic-usb 0003:0CCD:0028.0001: input: USB HID v1.00 Device [USB Audio]
on usb-0000:00:1d.1-2/input3

I can not comment about that "cannot find UAC_HEADER" error, but until
2.6.38 the card worked anyway.
With 2.6.39 chip->probing remains 1 on error exit, and any later ioctl
stops in snd_usb_autoresume with -ENODEV.

Signed-off-by: Thomas Pfaff <tpfaff@gmx.net>
Cc: <stable@kernel.org> [2.6.39+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 15:26:06 +02:00
Andy Shevchenko 49957f3966 ALSA: 6fire: don't use custom hex_to_bin()
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-23 15:18:52 +02:00
Daniele Guerrieri 14515a0829 ALSA: usb-audio: Added support for Roland UM-ONE midi-usb interface
Roland UM-ONE midi usb interface differs from Roland UM-1.

Signed-off-by: Daniele Guerrieri <d.guerrieri@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-16 08:31:45 +02:00
Daniel Mack c731bc96ad ALSA: snd-usb: move code from urb.c to endpoint.c
No code altered at this point, simply preparing for upcoming
refactorizations.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:07:03 +02:00
Daniel Mack e8e8babf56 ALSA: snd-usb: re-order code
Move code from endpoint.c into a new file called stream.c and rename
functions so that their names actually reflect what they're doing.

This way, endpoint.c will be available to functions that hold all the
endpoint logic.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:07:02 +02:00
Daniel Mack 358e2bd4a9 ALSA: snd-usb: re-order the Makefile
Sort its entries in alphabetical order.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 17:07:01 +02:00
Pierre-Louis Bossart 294c4fb8ab ALSA: usb: refine delay information with USB frame counter
Existing code only updates the audio delay when URBs were
submitted/retired. This can introduce an uncertainty of 8ms
on the number of samples played out with the default settings,
and a lot more when URBs convey more packets to reduce the
interrupt rate and power consumption.

This patch relies on the USB frame counter to reduce the
uncertainty to less than 2ms worst-case. The delay information
essentially becomes independent of the URB size and number of
packets. This should help applications like PulseAudio which
require accurate audio timing. Clemens Ladisch reported
a decrease of mplayer's A-V difference from nrpacks down to at
most 1ms.

Thanks to Clemens for also pointing out that the implementation
of frame counters varies between different HCDs. Only the
8 lowest-bits are used to estimate the delay.

Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[clemens: changed debug code]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:30:20 +02:00
Kristian Amlie 1ef0e0a053 ALSA: usb-audio: add Starr Labs USB MIDI support
Add support for Starr Labs USB MIDI devices such as the Z7S, which are
based on an FTDI serial UART chip.

Based on a patch by Daniel Mack.

Signed-off-by: Kristian Amlie <kristian@amlie.name>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-26 14:12:34 +02:00
Alan Stern 5b1b0b812a PM / Runtime: Add macro to test for runtime PM events
This patch (as1482) adds a macro for testing whether or not a
pm_message value represents an autosuspend or autoresume (i.e., a
runtime PM) event.  Encapsulating this notion seems preferable to
open-coding the test all over the place.

Signed-off-by: Alan Stern <stern@rowland.harvard.edu>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Rafael J. Wysocki <rjw@sisk.pl>
2011-08-19 23:49:48 +02:00
Takashi Iwai 9fcd0ab130 ALSA: usb-audio - Check the dB-range validity in the later read, too
When the initial check of dB-range failed due to the read error, try to
check again at the later read, too.  When an invalid dB range is found,
remove TLV flags and notify the mixer info change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-19 08:30:53 +02:00
Takashi Iwai 38b65190c6 ALSA: usb-audio - Fix missing mixer dB information
The recent fix for testing dB range at the mixer creation time seems
to cause regressions in some devices.  In such devices, reading the dB
info at probing time gives an error, thus both dBmin and dBmax are still
zero, and TLV flag isn't set although the later read of dB info succeeds.

This patch adds a workaround for such a case by assuming that the later
read will succeed.  In future, a similar test should be performed in a
case where a wrong dB range is seen even in the later read.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
2011-08-19 07:55:10 +02:00
Daniel Mack da6094ea7d ALSA: snd_usb_caiaq: track submitted output urbs
The snd_usb_caiaq driver currently assumes that output urbs are serviced
in time and doesn't track when and whether they are given back by the
USB core. That usually works fine, but due to temporary limitations of
the XHCI stack, we faced that urbs were submitted more than once with
this approach.

As it's no good practice to fire and forget urbs anyway, this patch
introduces a proper bit mask to track which requests have been submitted
and given back.

That alone however doesn't make the driver work in case the host
controller is broken and doesn't give back urbs at all, and the output
stream will stop once all pre-allocated output urbs are consumed. But
it does prevent crashes of the controller stack in such cases.

See http://bugzilla.kernel.org/show_bug.cgi?id=40702 for more details.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Matej Laitl <matej@laitl.cz>
Cc: Sarah Sharp <sarah.a.sharp@linux.intel.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-14 18:10:43 +02:00
Daniel Mack 15439bde3a ALSA: snd-usb-caiaq: Correct offset fields of outbound iso_frame_desc
This fixes faulty outbount packets in case the inbound packets
received from the hardware are fragmented and contain bogus input
iso frames. The bug has been there for ages, but for some strange
reasons, it was only triggered by newer machines in 64bit mode.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: William Light <wrl@illest.net>
Reported-by: Pedro Ribeiro <pedrib@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-10 20:05:47 +02:00
Kazutomo Yoshii c9c9e4e425 ALSA: usb-audio - Add quirk for BOSS Micro BR-80
Signed-off-by: Kazutomo Yoshii <kazutomo.yoshii@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-10 08:18:57 +02:00
Daniel Mack f4389489b5 ALSA: snd-usb-caiaq: Fix keymap for RigKontrol3
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Renato <naretobh@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-06 10:23:36 +02:00
Daniel Mack dac8f847c4 ALSA: snd-usb: Fix uninitialized variable usage
Purely cosmetic, but fixes the following build warning.

  CC [M]  sound/usb/quirks.o
sound/usb/quirks.c: In function ‘snd_usb_apply_boot_quirk’:
sound/usb/quirks.c:429:6: warning: ‘err’ may be used uninitialized in this function [-Wuninitialized]

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-06 10:22:58 +02:00
Miller Puckette 02651d1a97 ALSA: usb-audio - add quirk for Keith McMillen StringPort
Signed-off-by: Miller Puckette <msp@ucsd.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-05 08:27:02 +02:00
Daniel Mack 1faa5d07a9 ALSA: snd-usb: operate on given mixer interface only
When creating the mixers for an USB audio device, the current code looks
at the host interface stored in mixer->chip->ctrl_if. Change this and
rather keep a local pointer to the interface that was given when
snd_usb_create_mixer() was called.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Reported-by: Lean-Yves LENHOF <jean-yves@lenhof.eu.org>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-04 16:24:10 +02:00
Nicolai Krakowiak 60c961a9e1 ALSA: snd-usb: avoid dividing by zero on invalid input
Signed-off-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-04 16:24:06 +02:00
Clemens Ladisch 824818b148 ALSA: snd-usb: Accept UAC2 FORMAT_TYPE descriptors with bLength > 6
The Focusrite Scarlett 18i6 USB has them that way, which is probably a
bug. Anyway, the driver should simply ignore this fact.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-04 16:23:47 +02:00
David G Turner 6a6d822e12 ALSA: usb-audio - Add quirk for Roland / BOSS BR-800
Add support for Roland/BOSS BR-800 (0582:011e) to snd-usb-audio driver.

This allows playback and recording, which has been tested and found to
work. The third interface should be MIDI (MTC/SMPTE?) for DAW interface
and is set as per ME-25, but this has not been tested. SDHC card access
is already supported by usb-storage for Backup/Rhythm Editor/Wave
Convertor mode which should not conflict with this.

Signed-off-by: David G Turner <dgturner@iee.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-21 19:00:57 +02:00
Guillaume Pellerin 0f5733b0c8 ALSA: usb-audio - Add quirks for M-Audio Fast Track Pro and Quattro
This patch gives M-Audio Fast Track Pro and M-Audio Quattro quirks and
endpoints to boot and setup those devices with special options (digital
inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are
just adapted to match the new global M-Audio parameters.

Special configurations can be then loaded through a modprobe conf file.
For example, to set the 24 bits mode on the Fast Track Pro add
/etc/modprobe.d/fast_track_pro.conf :

    options snd_usb_audio   vid=0x763 pid=0x2012 device_setup=0x08

Here is a list of the possibilities in this example :
http://files.parisson.com/debian/fast-track-pro.conf

Signed-off-by: Guillaume Pellerin <yomguy@parisson.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-12 18:15:45 +02:00
Pavel Roskin 81b85b6bd9 ALSA: usb-audio: replace "void *" with more specific pointers
Signed-off-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-08 10:10:25 +02:00
Takashi Iwai 3409fcd1f7 Merge branch 'fix/hda' into topic/misc 2011-06-17 14:54:47 +02:00
Clemens Ladisch f4b1e98aa9 ALSA: firewire-speakers, oxygen, ua101: allow > 10 s periods
Since commit f2b3614cef (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 11:36:30 +02:00
Torsten Schenk 0ec5258d68 ALSA: 6fire - Fix signedness bug
Fixed remaining issues of the signedness bug discovered by Dan Carpenter.
A check was remaining that tests if unsigned rt->rate is >= 0.
Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates)
as invalid rate value and not -1.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-16 21:31:33 +02:00
Jesper Juhl 37f7ec38ea ALSA: 6fire: Fix double-free bug in usb6fire_fw_ezusb_upload()
We have a double-free bug in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload().
We already call release_firmware(fw) on line 258, so when we then do it
again after usb6fire_fw_ezusb_write() returns <0, we have a double-free.
Easily fixed by just removing the last call to release_firmware().

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-14 07:27:26 +02:00
Eric Lammerts 157186bc18 ALSA: usb - turn off de-emphasis in s/pdif for cm6206
CM6206: Turn off de-emphasis channel status bit in S/PDIF output.

Signed-off-by: Eric Lammerts <eric@lammerts.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-03 18:22:56 +02:00
Jesper Juhl bf0be0e951 ALSA: 6fire: Don't leak firmware in error path
One of the error paths in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload() neglects to free
the memory allocated for the firmware before returning, thus leaking the
memory.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-02 19:56:31 +02:00
Daniel Mack d5a0bf6cc5 ALSA: usb-audio: more control quirks for M-Audio FastTrack devices
Make use of the freshly introduced methods to re-use standard mixer
handling and add some controls that are hidden but implemented in a
standard conform way on M-Audio's FastTrack devices.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Original-code-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:37:08 +02:00
Daniel Mack 014950b013 ALSA: usb-audio: add new quirk type QUIRK_AUDIO_STANDARD_MIXER
This quirk type will let the driver assume that there is a standard
mixer on a given interface, or that a specific mixer quirks will handle
the device.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:36:59 +02:00
Daniel Mack 9e38658f70 ALSA: usb-audio: export snd_usb_feature_unit_ctl
In order to allow quirks functions to hook up to the standard feature
unit op tables, this patch exports a pointer to the struct that is used
internally.

That way, all the code handling the control can be kept private, and
external code can reference the symbol to re-use it.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:36:20 +02:00
Daniel Mack ef9d597089 ALSA: usb-audio: rework add_control_to_empty()
This patch renames add_control_to_empty() to snd_usb_mixer_add_control()
and exports it, so the quirks functions can make use of it.

Also, as "struct mixer_build" is private to mixer.c, rewrite the
function to take an argument of type "struct usb_mixer_interface"
instead.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:34:34 +02:00
Daniel Mack 5875c2cb76 ALSA: usb-audio: move assignment of chip->ctrl_intf
This is needed for upcoming changes to the quirks mechanism.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:34:19 +02:00
Takashi Iwai 02e5fbf622 Merge branch 'topic/misc' into for-linus 2011-05-22 10:01:29 +02:00
Mathieu Bouffard 7cdd8d7313 ALSA: usb-audio - Add support for USB X-Fi S51 Pro
USB X-Fi S51 Pro volume and mute from the volume knob on the unit.
Compiled and tested with 2.6.39-rc7-git12

Signed-off-by: Mathieu Bouffard <mbouffard@strangequarks.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 17:09:17 +02:00
Daniel Mack c91d9cda55 ALSA: usb-audio: handle "Fast Track Ultra" with USB_DEVICE_VENDOR_SPEC()
That way, the class compliant MIDI interface is also handled.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Grant Diffey <gdiffey@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 11:44:45 +02:00
Daniel Mack 3bc6fbc743 ALSA: usb-audio: assume valid clock
If the interface can't report a clock's validity, assume that it's
valid.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Vicente Joel <vicentejoel@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 11:44:44 +02:00
Daniel Mack 0ef283247a ALSA: usb-audio: add quirks for Roland GR-55
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Jeffrey Scott Flesher <jeffrey.scott.flesher@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 11:44:43 +02:00
Daniel Mack 56a9eb1f30 ALSA: usb-audio: Add quirk for KORG PANDORA PX5D MIDI interface
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Frédéric Jaume <frederic.jaume@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 11:44:42 +02:00
Daniel Mack 759e890f5c ALSA: usb-audio: remove invalid extra mixers for Komplete Audio 6
This was a flaw in the reading of the spec tables - Native Instrument's
"Komplete Audio 6" device has no such extra controls.

This patch also fixes the device name in two comments.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 11:44:42 +02:00
Daniel Mack ee95cb6121 ALSA: usb-audio: include format.h in format.c
Just in case a prototype changes, we'll be warned. This also fixes a
sparse warning.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 11:44:41 +02:00
Daniel Mack 60ed286a61 ALSA: usb-audio: make hwc_debug a noop in case HW_CONST_DEBUG is not set
Just defining it to nothing is dangerous as it can alter the code
execution flow, for example when used in as only function in a
conditional code block.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-18 11:44:35 +02:00
Wolfgang Breyha 8129e79ed7 ALSA: usb-audio - Terratec Aureon 7.1 USB ID as C-Media cm6206 quirks
This patch adds support for the Terratec Aureon 7.1 USB which uses a
C-Media cm6206 and needs all the quirks already found in the past.

Signed-off-by: Wolfgang Breyha <wbreyha@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 18:22:41 +02:00
Takashi Iwai 59bb7f0eeb ALSA: usb-audio - Don't expose broken dB ranges
Some crappy USB-audio devices give broken dB ranges, e.g. both min and max
are 0dB.  This confuses the volume control that prefers dB expression such
as alsactl or PulseAudio.  In such a case, it's much better not to expose
the broken dB information.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-28 09:58:43 +02:00
Daniel Mack 8ae9572b5b ALSA: 6fire: use the kernel's built-in bit reverse table
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-26 12:26:12 +02:00
Daniel Mack 9cdc352936 ALSA: usb-audio: Add quirks for Audio Kontrol 6
This new device by Native Instruments is also compliant to the USB
standard v2.0, but hides this detail at when connected.

It needs the same boot quirks than other models, and also has two
non-class-compliant mixer controls.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-14 12:06:02 +02:00
Linus Torvalds 4263a2f1da Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Don't query connections for widgets have no connections
  ALSA: HDA: Fix single internal mic on ALC275 (Sony Vaio VPCSB1C5E)
  ALSA: hda - HDMI: Fix MCP7x audio infoframe checksums
  ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
  ALSA: HDA: Fix dock mic for Lenovo X220-tablet
  ASoC: format_register_str: Don't clip register values
  ASoC: PXA: Fix oops in __pxa2xx_pcm_prepare
  ASoC: zylonite: set .codec_dai_name in initializer
2011-04-10 09:56:10 -07:00
Tarek Soliman 49c039f071 ALSA: usb-audio: define another USB ID for a buggy USB MIDI cable
There are many USB MIDI cables out there that have buggy
firmware that reports it can do more than 4 bytes in a
packet when they can only properly handle 4

This patch adds the ID of yet another one of those cables

Signed-off-by: Tarek Soliman <tarek@bashasoliman.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-06 08:05:30 +02:00
Torsten Schenk 2475b0d407 ALSA: 6fire - Add support of digital-thru mixer
Digital Thru mixer element added (device can act as converter optical<->coax)

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-04 12:26:21 +02:00
Torsten Schenk b84610b95f ALSA: 6fire - Improve firmware loader
Firmware loader: magical device bytes check updated (accepts all device
versions now and accepts possibly loaded firmware, if it is knowing to
be working)

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-04 12:25:50 +02:00
Torsten Schenk 58c54fa47f ALSA: 6fire - Add support for S32_LE format
Added support for sample format s32_le.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-04 12:25:13 +02:00
Torsten Schenk e220fa3bf5 ALSA: 6fire - Fix pcm rate assignment
Completion of signedness bug for pcm_runtime.rate: variable will never
get assigned a negative value now.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-04 12:24:53 +02:00
Torsten Schenk 78caf66cb5 ALSA: 6fire - Update kernel configuration
Kernel configuration updated:
- experimental dependency removed
- description updated

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-04-04 12:23:28 +02:00
Lucas De Marchi 25985edced Fix common misspellings
Fixes generated by 'codespell' and manually reviewed.

Signed-off-by: Lucas De Marchi <lucas.demarchi@profusion.mobi>
2011-03-31 11:26:23 -03:00
Keith A. Milner cb6f4b55f5 ALSA: usb-audio - Support for Boss JS-8 Jam Station
Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-23 22:51:38 +01:00
Clemens Ladisch e19869204f ALSA: usb-audio: add Cakewalk UM-1G support
Add a quirk for the Cakewalk UM-1G USB MIDI interface in
"advanced driver" mode.  (It already works in standard mode.)

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-23 22:49:19 +01:00
Takashi Iwai 3ffc1222bd ALSA: usb - Remove trailing spaces from USB card name strings
Some USB devices give trailing spaces in strings returned from
usb_string().  This confuses the automatic card-id creation, resulting
always in "default".
This patch fixes the behavior by removing trailing spaces.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-21 12:32:44 +01:00
Oliver Neukum 88a8516a21 ALSA: usbaudio: implement USB autosuspend
Devices are autosuspended if no pcm nor midi channel is open
Mixer devices may be opened. This way they are active when
in use to play or record sound, but can be suspended while
users have a mixer application running.

[Small clean-ups using static inline by tiwai]

Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11 14:59:29 +01:00
Oliver Neukum edf7de31c2 ALSA: usbaudio: fix suspend/resume
- ESHUTDOWN must be correctly handled
- the optional interrupt endpoint's URB must be stopped and restarted

Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-11 14:51:51 +01:00
Takashi Iwai cc99a0861f Merge branch 'fix/misc' into topic/misc 2011-03-11 14:48:09 +01:00
Alexey Fisher bc3a8a01de ALSA: usb-audio - Add "cval->res = 384" quirk for Logitech Webcam C600
One more affected devices: Logitech Webcam C600 (046d:0808)

Volume range before quirk is 6400, after (also real) is 16.

Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10 16:48:21 +01:00
Alexey Fisher 80acefff3b ALSA: usb-audio - Add volume range check and warn if it too big
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-10 16:47:56 +01:00
Takashi Iwai 382225e62b ALSA: usb-audio: fix oops due to cleanup race when disconnecting
When a USB audio device is disconnected, snd_usb_audio_disconnect()
kills all audio URBs.  At the same time, the application, after being
notified of the disconnection, might close the device, in which case
ALSA calls the .hw_free callback, which should free the URBs too.

Commit de1b8b93a0 "[ALSA] Fix hang-up at disconnection of usb-audio"
prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that
resulted from this race, but this introduced another race because the
URB callbacks could now be executed after snd_usb_hw_free() has
returned, and try to access already freed data.

Fix the first race by introducing a mutex to serialize the disconnect
callback and all PCM callbacks that manage URBs (hw_free and hw_params).

Reported-and-tested-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
Cc: <stable@kernel.org>
[CL: also serialize hw_params callback]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-02-23 08:15:43 +01:00
Takashi Iwai eaae55dac6 ALSA: caiaq - Fix possible string-buffer overflow
Use strlcpy() to assure not to overflow the string array sizes by
too long USB device name string.

Reported-by: Rafa <rafa@mwrinfosecurity.com>
Cc: stable <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-02-14 22:50:46 +01:00
Daniel Mack 3347b26cab ALSA: usb-audio: reconstruct some dispatcher functions to use switch-case
The number of cases has increased so use switch-case rather than
if-statements.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-02-14 17:11:12 +01:00
Daniel Mack 54a8c500d5 ALSA: usb-audio: add support for Native Instruments MK2 devices
The MK2 generation of Native Instruments' sound cards are in fact
compliant to the USB audio standard of version 2 and other approved USB
standards. However, they come up as vendor-specific device when first
connected but can be told to come up with a new set of descriptors
upon their next enumeration. The interfaces announced by the new
descriptors will be handled by the kernel's class drivers. This is done
by issuing a vendor specific device request and sending the device to
reset.

There are also some vendor-specific USB requests for some mixer elements
that can't be exported in a standard compliant way. The driver now
supports them with quirks handling mechanisms.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-02-14 17:10:57 +01:00
Daniel Mack df8d81a32f ALSA: snd-usb-caiaq: Add support for Traktor Audio 2
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-02-14 17:10:45 +01:00
Joseph Teichman 1cdfa9f34a ALSA: usbaudio - Enable the E-MU 0204 USB
Signed-off-by: Joseph Teichman <josteich@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-02-08 08:36:04 +01:00
Dan Carpenter 8121d91c02 ALSA: USB: 6fire: signedness bug in usb6fire_pcm_prepare()
rt->rate is an unsigned char so it's never equal to -1.  It's not a huge
problem because the invalid rate is caught inside the call to
usb6fire_pcm_set_rate() which returns -EINVAL.  But if we fix the test
then it prints out the correct error message so that's good.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-02-07 18:30:10 +01:00
Torsten Schenk c6d43ba816 ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB
What is working: Everything except SPDIF
- Hardware Master volume
- PCM 44-192kHz@24 bits, 6 channels out, 4 channels in (analog)
- MIDI in/out
- firmware loading after cold start
- phono/line switching

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-24 18:45:30 +01:00
Karsten Wiese cdce2db74e ALSA: snd-usb-us122l: Fix missing NULL checks
Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl().
Wake up poll waiters before returning from usb_stream_hwdep_ioctl().

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-11 19:50:28 +01:00
Karsten Wiese 921eebdc18 ALSA: snd-usb-us122l: Fix MIDI output
The US-122L always reads 9 bytes per urb unless they are set to 0xFD.

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-01-11 19:48:58 +01:00