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>

Commit 88a8516a (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 f5f16541 (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: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NCalvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

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

Show the error code returned from the USB subsystem in
the debug messages.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

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

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

Replace the endianness conversions with the kernel-wide swabbing macros
in get/set_sample_rate_v2.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Correct spelling of snd_usb_endpoint_implict_feedback_sink in all
occurances.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Put EXPORT_SYMBOLS directly under the exported function.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Minor style fix, following a general code style in the kernel.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Change occurances of list_for_each into list_for_each_entry where
applicable.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>

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: NTorstein Hegge <hegge@resisty.net>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Reported-and-tested-by: NRodolfo Thomazelli <pe.soberbo@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTorstein Hegge <hegge@resisty.net>
Acked-by: NDaniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

"Playback Design" products need a 50ms delay after setting the USB
interface.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NAndreas Koch <andreas@akdesigninc.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Reported-by: NAndreas Koch <andreas@akdesigninc.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This field may use up to 32 bits, so it should be handled as unsigned
int.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NAndreas Koch <andreas@akdesigninc.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMark Hills <mark@xwax.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMark Hills <mark@xwax.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDave Helstroom <helstroom@google.com>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

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

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

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

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: NMatt Gruskin <matthew.gruskin@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NClemens Ladisch <clemens@ladisch.de>
Cc: 2.6.37+ <stable@vger.kernel.org>

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

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: NAntonio Ospite <ao2@amarulasolutions.com>
Acked-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Commit 23caaf19 (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: NFlorian Hanisch <fhanisch@uni-potsdam.de>
Tested-by: NMatthew Robbetts <wingfeathera@gmail.com>
Tested-by: NMichael Beer <beerml@sigma6audio.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: 2.6.35+ <stable@vger.kernel.org>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NStratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add names of the clock sources for the M-Audio Fast Track
C400.
Signed-off-by: NEldad Zack <eldad@fogrefinery.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

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

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/1095315Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NJens Axboe <axboe@kernel.dk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>