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: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDylan Reid <dgreid@chromium.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NMarkus Trippelsdorf <markus@trippelsdorf.de>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Commit 68e67f40 ("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: NDaniel Mack <zonque@gmail.com>
Reported-by: NRobert M. Albrecht <linux@romal.de>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Commit e9ba389c ("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: NDaniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
Reported-and-tested-by: NPhilipp Dreimann <philipp@dreimann.net>
Reported-by: NJoseph Salisbury <joseph.salisbury@canonical.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NTakashi Iwai <tiwai@suse.de>

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

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: NDaniel Mack <zonque@gmail.com>
Reported-and-tested-by: <ben@b1c1l1.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: N"Alexander E. Patrakov" <patrakov@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Set some substream struct members to make the proc interface code work
again.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NFelix Homann <linuxaudio@showlabor.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NDaniel Mack <zonque@gmail.com>
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>

Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NFelipe Balbi <balbi@ti.com>
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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>

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: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
[clemens: changed debug code]
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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: NOliver Neukum <oneukum@suse.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

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 de1b8b93 "[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: NPierre-Louis Bossart <pierre-louis.bossart@intel.com>
Cc: <stable@kernel.org>
[CL: also serialize hw_params callback]
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

sound/usb/pcm.c::snd_usb_pcm_check_knot() fails to check the return value
from kmalloc() and may end up dereferencing a null pointer.
The patch below (compile tested only) should take care of that little
problem.
Signed-off-by: NJesper Juhl <jj@chaosbits.net>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

There are two USB Audio Class specifications (v1 and v2), but neither of
them clearly defines the feedback format for high-speed UAC v1 devices.
Add to this whatever the Creative and M-Audio firmware writers have been
smoking, and it becomes impossible to predict the exact feedback format
used by a particular device.

Therefore, automatically detect the feedback format by looking at the
magnitude of the first received feedback value.

Also, this allows us to get rid of some special cases for E-Mu devices.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
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>

This is V2 of the patch, after feedback from Clemens and Daniel.

This patch adds SuperSpeed support to the USB drivers under sound/. It adds
tests for USB_SPEED_SUPER to the appropriate places that check for the USB
speed.

This patch has been tested with our SS USB3 device emulating a set of Yamaha
speakers and a Logitech microphone, but with the descriptors modified to add
USB3 support. It has also been tested with the real speakers and microphone,
to make sure that USB2 devices still work.
Signed-off-by: NPaul Zimmerman <paulz@synopsys.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NUwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: NJiri Kosina <jkosina@suse.cz>

Audio devices which comply to the UAC2 standard can export complex clock
topologies in its descriptors and set up links between them.

The entities that are defined are

 - clock sources, which define the end-leafs.
 - clock selectors, which act as switch to select one out of many
   possible clocks sources.
 - clock multipliers, which have an input clock source, and act as clock
   source again. They can be used to derive one clock from another.

All sample rate changes, clock validity queries and the like must go to
clock source elements, while clock selectors and multipliers can be used
as terminal clock source.

The following patch adds a parser for these elements and functions to
iterate over the tree and find the leaf nodes (clock sources).

The samplerate set functions were moved to the new clock.c file.
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This request is again handled differently in comparison to UAC1.
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NStephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

- Split the audio.h file in two to clearly denote the differences
  between the standards.
- Add many more defines to audio-v2.h. Most of them are not currently
  used.
- Replaced a magic value with a proper define
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Acked-by: NGreg Kroah-Hartman <gregkh@suse.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Sample rate setting is done with a 4-byte long class request that
addresses the interface.
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

In preparation for USB audio 2.0 support, change the audioformat
structure so that it uses a bitmask to specify possible formats.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The snd_usb_substream::format field actually contains the index of the
current alternate setting, so rename it to altset_idx to avoid
confusion.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Clean up the usb audio driver by factoring out a lot of functions to
separate files. Code for procfs, quirks, urbs, format parsers etc all
got a new home now.

Moved almost all special quirk handling to quirks.c and introduced new
generic functions to handle them, so the exceptions do not pollute the
whole driver.

Renamed usbaudio.c to card.c because this is what it actually does now.
Renamed usbmidi.c to midi.c for namespace clarity.
Removed more things from usbaudio.h.

The non-standard drivers were adopted accordingly.
Signed-off-by: NDaniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>