Commit Graph

782 Commits

Author SHA1 Message Date
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