Following general kernel style.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Code block does not compile when enabled.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Commit 8f898e92 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: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NPrzemek Rudy <prudy1@o2.pl>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NClemens Ladisch <clemens@ladisch.de>

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: NClemens Ladisch <clemens@ladisch.de>

Remove all quirks that are no longer needed now that the generic Roland
quirks can handle the vendor-specific descriptors correctly.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

Signed-off-by: NClemens Ladisch <clemens@ladisch.de>

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: NClemens Ladisch <clemens@ladisch.de>

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: NClemens Ladisch <clemens@ladisch.de>

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: NClemens Ladisch <clemens@ladisch.de>

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: NAntonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NAntonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For USB devices it's not necessary to allocate physically contiguous
buffers.
Signed-off-by: NAntonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

For USB devices it's not necessary to allocate physically contiguous
buffers.
Signed-off-by: NAntonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The snd_card_used variable is only read but never written, remove it.
Signed-off-by: NAntonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NDave Jones <davej@redhat.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

USB_QUEUE_BULK isn't defined any more.
Signed-off-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Just like the previous fix for LogitechHD Webcam c270 in commit
11e7064f, 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: NTakashi Iwai <tiwai@suse.de>

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: NJeremy Rosen <jeremy.rosen@openwide.fr>
Tested-by: NJeremy Rosen <jeremy.rosen@openwide.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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=821735Reported-and-tested-by: NCristian Rodríguez <crrodriguez@opensuse.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

... instead of applying to all interfaces.

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

Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Commit 927c9423 (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: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTorsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NB. Zhang <bb.zhang@free.fr>
Signed-off-by: NTorstein Hegge <hegge@resisty.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NJoe Rayhawk <jrayhawk@fairlystable.org>
Cc: <stable@vger.kernel.org> # 3.8 only
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds a USB quirk for the Yamaha THR10C amp.
Signed-off-by: NTrulan Martin <trulanm@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds a USB quirk for the Yamaha THR5A amp.
Signed-off-by: NTrulan Martin <trulanm@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds a USB quirk for the Yamaha THR10 amp.
Signed-off-by: NTrulan Martin <trulanm@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Schürmann <daschuer@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Reported-and-tested-by: NTorstein Hegge <hegge@resisty.net>
Reported-and-tested-by: NYves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Schürmann <daschuer@mixxx.org>
Acked-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.

The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.

To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.

The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>