Currently, USB-audio driver allocates the PCM buffer via vmalloc(), as
this serves merely as an intermediate buffer that is copied to each
URB transfer buffer.  This works well in general on x86, but on some
archs this may result in cache coherency issues when mmap is used.
OTOH, it works also on such arch unless mmap is used.

This patch is a step for mitigating the inconvenience; a new module
option "use_vmalloc" is provided so that user can choose to allocate
the DMA coherent buffer instead of the existing vmalloc buffer.
The drawback is that it'd be the standard dma_alloc_coherent() calls
and the system would require contiguous pages on non-x86 archs.

Note that it's a global option and not dynamically switchable since
the buffer is pre-allocated at the probe time.  In theory, it's
possible to be switchable, but it'd be trickier and racier.

As default use_vmalloc option is set to true, so that the old behavior
is kept.  For allowing the coherent mmap on ARM or MIPS, pass
use_vmalloc=0 option explicitly.
Reported-and-tested-by: NDaniel Danzberger <daniel@dd-wrt.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

UAC3 channel map is created during interface parsing,
and in some cases was not freed in failure paths.
Reported-by: NDan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

bmAtributes offset doesn't exist in the UAC3 CS_EP descriptor.
Hence, checking for pitch control as if it was UAC2 doesn't make
any sense. Use the defined UAC3 offsets instead.

Fixes: 9a2fe9b8 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Signed-off-by: NJorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Recently released USB Audio Class 3.0 specification
contains BADD (Basic Audio Device Definition) document
which describes pre-defined UAC3 configurations.

BADD support is mandatory for UAC3 devices, it should be
implemented as a separate USB device configuration.
As per BADD document, class-specific descriptors
shall not be included in the Device’s Configuration
descriptor ("inferred"), but host can guess them
from BADD profile number, number of endpoints and
their max packed sizes.

This patch adds support of all BADD profiles from the spec
Signed-off-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: NJorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Offload snd_usb_parse_audio_interface() function
which became quite long after adding UAC3 spec support.

Move class-specific parts of uac3 parsing to separate
function which now produce audioformat structure that
is ready to be fed to snd_usb_add_audio_stream().
Signed-off-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Offload snd_usb_parse_audio_interface() function
which became quite long after adding UAC3 spec support.

Move class-specific parts of uac1/2 parsing to separate
function which now produce audioformat structure that
is ready to be fed to snd_usb_add_audio_stream().

This also broke Blue Microphones workaround (which
relies on audioformat decoded from previous altsetting)
into two parts: prepare quirk flag analyzing previous
altsetting then use it with current altsetting.
Signed-off-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Offload snd_usb_parse_audio_interface() function which
became quite long after adding UAC3 spec support.

Move audioformat allocation and initialization
into separate function, this will make easier
future refactoring.
Attributes left in the original func because it'll
be used for UAC3 BADD profiles suport in the future

There is no functional change.
Signed-off-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The channel mapping is defined by bChRelationship, not bChPurpose.

Fixes: 9a2fe9b8 ("ALSA: usb: initial USB Audio Device Class 3.0 support")
Reviewed-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: NMichael Drake <michael.drake@codethink.co.uk>
Signed-off-by: NJorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Recently released USB Audio Class 3.0 specification
introduces many significant changes comparing to
previous versions, like
 - new Power Domains, support for LPM/L1
 - new Cluster descriptor
 - changed layout of all class-specific descriptors
 - new High Capability descriptors
 - New class-specific String descriptors
 - new and removed units
 - additional sources for interrupts
 - removed Type II Audio Data Formats
 - ... and many other things (check spec)

It also provides backward compatibility through
multiple configurations, as well as requires
mandatory support for BADD (Basic Audio Device
Definition) on each ADC3.0 compliant device

This patch adds initial support of UAC3 specification
that is enough for Generic I/O Profile (BAOF, BAIF)
device support from BADD document.
Signed-off-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Reviewed-by: NGreg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Offload USB audio interface parsing function by
moving quirks to a specially designed location (quirks.c)
Signed-off-by: NRuslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Omit an extra message for a memory allocation failure in these functions.

This issue was detected by using the Coccinelle software.

Link: http://events.linuxfoundation.org/sites/events/files/slides/LCJ16-Refactor_Strings-WSang_0.pdfSigned-off-by: NMarkus Elfring <elfring@users.sourceforge.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Unfortunately, this patch caused several regressions at au0828 and
snd-usb-audio, like this one:
	https://bugzilla.kernel.org/show_bug.cgi?id=115561

It also showed several troubles at the MC core that handles pretty
poorly the memory protections and data lifetime management.

So, better to revert it and fix the core before reapplying this
change.

This reverts commit aebb2b89 ("[media] sound/usb: Use Media
Controller API to share media resources")'
Signed-off-by: NMauro Carvalho Chehab <mchehab@osg.samsung.com>

create_fixed_stream_quirk(), snd_usb_parse_audio_interface() and
create_uaxx_quirk() functions allocate the audioformat object by themselves
and free it upon error before returning. However, once the object is linked
to a stream, it's freed again in snd_usb_audio_pcm_free(), thus it'll be
double-freed, eventually resulting in a memory corruption.

This patch fixes these failures in the error paths by unlinking the audioformat
object before freeing it.

Based on a patch by Takashi Iwai <tiwai@suse.de>

[Note for stable backports:
 this patch requires the commit 902eb7fd ('ALSA: usb-audio: Minor
 code cleanup in create_fixed_stream_quirk()')]

Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1283358Reported-by: NRalf Spenneberg <ralf@spenneberg.net>
Cc: <stable@vger.kernel.org> # see the note above
Signed-off-by: NVladis Dronov <vdronov@redhat.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Change ALSA driver to use Media Controller API to share media resources
with DVB and V4L2 drivers on a AU0828 media device. Media Controller
specific initialization is done after sound card is registered. ALSA
creates Media interface and entity function graph nodes for Control,
Mixer, PCM Playback, and PCM Capture devices.

snd_usb_hw_params() will call Media Controller enable source handler
interface to request the media resource. If resource request is
granted, it will release it from snd_usb_hw_free(). If resource is
busy, -EBUSY is returned.

Media specific cleanup is done in usb_audio_disconnect().
Signed-off-by: NShuah Khan <shuahkh@osg.samsung.com>
Acked-by: NTakashi Iwai <tiwai@suse.de>
Signed-off-by: NMauro Carvalho Chehab <mchehab@osg.samsung.com>

For better readability, use list_for_each_entry_continue_reverse()
in have_dup_chmap().
Signed-off-by: NGeliang Tang <geliangtang@163.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The Zoom R16/24 have a nonstandard playback format where each isochronous
packet contains a length descriptor in the first four bytes. (Curiously,
capture data does not contain this and requires no quirk.)

The quirk involves adding the extra length descriptor whenever outgoing
isochronous packets are generated, both in pcm.c (outgoing audio) and
endpoint.c (silent data).

In order to make the quirk as unintrusive as possible, for
pcm.c:prepare_playback_urb(), the isochronous packet descriptors are
initially set up in the same way no matter if the quirk is enabled or not.
Once it is time to actually copy the data into the outgoing packet buffer
(together with the added length descriptors) the isochronous descriptors
are adjusted in order take the increased payload length into account.

For endpoint.c:prepare_silent_urb() it makes more sense to modify the
actual function, partly because the function is less complex to start with
and partly because it is not as time-critical as prepare_playback_urb()
(whose bulk is run with interrupts disabled), so the (minute) additional
time spent in the non-quirk case is motivated by the simplicity of having
a single function for all cases.

The quirk is controlled by the new tx_length_quirk member in struct
snd_usb_substream and struct snd_usb_audio, which is conveyed to pcm.c
and endpoint.c from quirks.c in a similar manner to the txfr_quirk member
in the same structs.

In contrast to txfr_quirk however, the quirk is enabled directly in
quirks.c:create_standard_audio_quirk() by checking the USB ID in that
function. Another option would be to introduce a new
QUIRK_AUDIO_ZOOM_INTERFACE or somesuch, which would have made the quirk
very plain to see in the quirk table, but it was felt that the additional
code needed to implement it this way would just make the implementation
more complex with no real gain.

Tested with a Zoom R16, both by doing capture and playback separately
using arecord and aplay (8 channel capture and 2 channel playback,
respectively), as well as capture and playback together using Ardour, as
well as Audacity and Qtractor together with jackd.

The R24 is reportedly compatible with the R16 when used as an audio
interface. Both devices share the same USB ID and have the same number of
inputs (8) and outputs (2). Therefore "R16/24" is mentioned throughout the
patch.

Regression tested using an Edirol UA-5 in both class compliant (16-bit)
and "advanced" (24 bit, forces the use of quirks) modes.
Signed-off-by: NRicard Wanderlof <ricardw@axis.com>
Tested-by: NPanu Matilainen <pmatilai@laiskiainen.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

New PCMs will now be added to the end of the chip's PCM list instead of to the
front. This changes the way streams are combined so that the first capture
stream will now be merged with the first playback stream instead of the last.

This fixes a problem with ASUS U7. Cards with one playback stream and cards
without capture streams should be unaffected by this change.

Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to
swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf
Signed-off-by: NJohan Rastén <johan@oljud.se>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Convert with dev_err() and co from snd_printk(), etc.
As there are too deep indirections (e.g. ep->chip->dev->dev),
a few new local macros, usb_audio_err() & co, are introduced.

Also, the device numbers in some messages are dropped, as they are
shown in the prefix automatically.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

According to USB Audio spec v2 bits 25 and 26 of bmChannelConfig are
"Back Left of Center - BLC" and "Back Right of Center - BRC",
respectively.

They are currently assigned to ALSA channels BLC/BRC. However, the ALSA
BLC/BRC are actually the rather nonsensical "bottom left center" and
"bottom right center", so the channels will be assigned wrongly. The
comments in the USB code are also similarly wrong, so this is not
readily apparent without looking at the actual specification.

Fix the channel mapping by mapping bits 25 and 26 to RLC (Rear Left
Center) and RRC (Rear Right Center), respectively, instead.
Signed-off-by: NAnssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In case the channel count of the input terminal is not the same as
the channel count of the streaming descriptor, the channel config of
the input terminal can not be trusted. Instead fall back to a default
(guessed) channel map.

This was found on a Logitech USB Headset.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The channel config from the streaming descriptor is probably a
better indicator of the channel map than the input terminal.
Use the input terminal's channel map as fallback only.
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

If wChannelconfig is given for some formats but not others, userspace
might not be able to set the channel map.

This is RFC because I'm not sure what the best behaviour is - to guess
the channel map from the given number of channels (it's quite likely
that one channel is MONO and two channels is FL FR), or just to supply
UNKNOWN for all channels.

But the complete lack of channel map for a format leads userspace to
believe that the format is not available at all. Or am I
misunderstanding how this should be used?
Signed-off-by: NDavid Henningsson <david.henningsson@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.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>

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>

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>

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>

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>

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>

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: NTakashi Iwai <tiwai@suse.de>

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: NMatthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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-g71fae7e7 #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: NJamie Heilman <jamie@audible.transient.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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

No code altered at this point, simply preparing for upcoming
refactorizations.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Reported-by: NNicolai Krakowiak <nicolai.krakowiak@gmail.com>
Cc: stable@kernel.org
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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.confSigned-off-by: NGuillaume Pellerin <yomguy@parisson.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The Audio Class v2 support code in 2.6.35 added checks for the
bInterfaceProtocol field.  However, there are devices (usually those
detected by vendor-specific quirks) that do not have one of the
predefined values in this field, which made the driver reject them.

To fix this regression, restore the old behaviour, i.e., assume that
a device with an unknown bInterfaceProtocol field (other than
UAC_VERSION_2) has more or less UAC-v1-compatible descriptors.

[compile warning fixes by tiwai]
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The M-Audio Fast Track Ultra series devices did not play sound correctly
at 44.1/88.2 kHz. Changing the output endpoint attribute to adaptive
fixes this.
Signed-off-by: NFelix Homann <fexpop@web.de>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

As the control interface is now carried in struct snd_usb_audio, we can
simplify the API a little and also drop the private ctrlif field from
struct usb_mixer_interface.

Also remove a left-over function prototype in pcm.h.
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Get rid of the last occurances of _v1 suffixes, and move the version
number right after the "uac" string. Now things are consitent again.

Sorry for the forth and back, but it just looks much nicer this way.
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>