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>
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>
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>
For implicit feedback endpoints, the number of bytes for each packet
is matched by the corresponding synchronizing endpoint.
The size is calculated by taking the actual size and dividing it by
the stride - currently by the endpoint's stride, but we should use the
synchronization source's stride.
This is evident when the number of channels differ between the
synchronization source and the implicitly fed-back endpoint, as with
M-Audio Fast Track C400 - the synchronization source (capture)
has 4 channels, while the implicit feedback mode endpoint has 6.
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As we are stopping the endpoints asynchronously now, it's better to
trigger the stop of both data and sync endpoints and wait for pending
stopping operations, instead of the sequential trigger-and-wait
procedure.
So the wait argument in snd_usb_endpoint_stop() is dropped, and it's
expected that the caller synchronizes explicitly by calling
snd_usb_endpoint_sync_pending_stop(). (Actually there is only one
place calling this, so it was safe to change.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
For further code simplification, drop the conditional call for
usb_kill_urb() with can_wait argument in deactivate_urbs(), and use
only usb_unlink_urb() and wait_clear_urbs() pairs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reduce the redundant arguments for snd_usb_endpoint_start() and
snd_usb_endpoint_stop(). Also replaced from int to bool.
No functional changes by this commit.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The async unlink behavior has been working over years. The option was
provided only as a workaround for 2.4.x kernel. Let's get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use bitmap_weight to count the total number of bits set in bitmap.
Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are bug reports of a crash with USB-audio devices when PCM
prepare is performed immediately after the stream is stopped via
trigger callback. It turned out that the problem is that we don't
wait until all URBs are killed.
This patch adds a new function to synchronize the pending stop
operation on an endpoint, and calls in the prepare callback for
avoiding the crash above.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=49181
Reported-and-tested-by: Artem S. Tashkinov <t.artem@lycos.com>
Cc: <stable@vger.kernel.org> [v3.6]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support of
D3 clock-stop. Also changing the power_save option in sysfs kicks
off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in HD-audio
are continued cleanups and standardization for the generic auto
parser and bug fixes (HBR, device-specific fixups), in addition to
the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode.
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Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This contains pretty many small commits covering fairly large range of
files in sound/ directory. Partly because of additional API support
and partly because of constantly developed ASoC and ARM stuff.
Some highlights:
- Introduced the helper function and documentation for exposing the
channel map via control API, as discussed in Plumbers; most of PCI
drivers are covered, will follow more drivers later
- Most of drivers have been replaced with the new PM callbacks (if
the bus is supported)
- HD-audio controller got the support of runtime PM and the support
of D3 clock-stop. Also changing the power_save option in sysfs
kicks off immediately to enable / disable the power-save mode.
- Another significant code change in HD-audio is the rewrite of
firmware loading code. Other than that, most of changes in
HD-audio are continued cleanups and standardization for the generic
auto parser and bug fixes (HBR, device-specific fixups), in
addition to the support of channel-map API.
- Addition of ASoC bindings for the compressed API, used by the
mid-x86 drivers.
- Lots of cleanups and API refreshes for ASoC codec drivers and
DaVinci.
- Conversion of OMAP to dmaengine.
- New machine driver for Wolfson Microelectronics Bells.
- New CODEC driver for Wolfson Microelectronics WM0010.
- Enhancements to the ux500 and wm2000 drivers
- A new driver for DA9055 and the support for regulator bypass mode."
Fix up various arm soc header file reorg conflicts.
* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
ALSA: hda - Add new codec ALC283 ALC290 support
ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
ALSA: hda - fix indices on boost volume on Conexant
ALSA: aloop - add locking to timer access
ALSA: hda - Fix hang caused by race during suspend.
sound: Remove unnecessary semicolon
ALSA: hda/realtek - Fix detection of ALC271X codec
ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
ALSA: hda - make a generic unsol event handler
ASoC: codecs: Add DA9055 codec driver
ASoC: eukrea-tlv320: Convert it to platform driver
ALSA: ASoC: add DT bindings for CS4271
ASoC: wm_hubs: Ensure volume updates are handled during class W startup
ASoC: wm5110: Adding missing volume update bits
ASoC: wm5110: Add OUT3R support
ASoC: wm5110: Add AEC loopback support
ASoC: wm5110: Rename EPOUT to HPOUT3
ASoC: arizona: Add more clock rates
ASoC: arizona: Add more DSP options for mixer input muxes
...
Also fix the calls to next_packet_size() for the pause case. This was
missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size").
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de>
Cc: stable@kernel.org
[ Taking directly because Takashi is on vacation - Linus ]
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Playback Designs' USB devices have some hardware limitations on their
USB interface. In particular:
- They need a 20ms delay after each class compliant request as the
hardware ACKs the USB packets before the device is actually ready
for the next command. Sending data immediately will result in buffer
overflows in the hardware.
- The devices send bogus feedback data at the start of each stream
which confuse the feedback format auto-detection.
This patch introduces a new quirks hook that is called after each
control packet and which adds a delay for all devices that match
Playback Designs' USB VID for now.
In addition, it adds a counter to snd_usb_endpoint to drop received
packets on the floor. Another new quirks function that is called once
an endpoint is started initializes that counter for these devices on
their sync endpoint.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com>
Supported-by: Demian Martin <demianm_1@yahoo.com>
Signed-off-by: Takashi 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: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit e9ba389c5 ("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: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi 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: Takashi 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: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Philipp Dreimann <philipp@dreimann.net>
Reported-by: Joseph Salisbury <joseph.salisbury@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
sound/usb/endpoint.c: In function 'queue_pending_output_urbs':
sound/usb/endpoint.c:298: warning: 'packet' may be used uninitialized in this function
Cc: Daniel Mack <zonque@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ep->fill_max is a 1 bit flag, thus it has to be boolean.
sound/usb/endpoint.c: In function 'snd_usb_endpoint_set_params':
sound/usb/endpoint.c:785: warning: overflow in implicit constant conversion
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Document the new streaming code and some of the functions so that
contributers can catch up easier.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi 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: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds a new generic streaming logic for audio over USB.
It defines a model (snd_usb_endpoint) that handles everything that
is related to an USB endpoint and its streaming. There are functions to
activate and deactivate an endpoint (which call usb_set_interface()),
and to start and stop its URBs. It also has function pointers to be
called when data was received or is about to be sent, and pointer to
a sync slave (another snd_usb_endpoint) that is informed when data has
been received.
A snd_usb_endpoint knows about its state and implements a refcounting,
so only the first user will actually start the URBs and only the last
one to stop it will tear them down again.
With this sort of abstraction, the actual streaming is decoupled from
the pcm handling, which makes the "implicit feedback" mechanisms easy to
implement.
In order to split changes properly, this patch only adds the new
implementation but leaves the old one around, so the the driver doesn't
change its behaviour. The switch to actually use the new code is
submitted separately.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With some buggy devices, the usb-audio driver may give "frame xxx active"
kernel messages too often. Better to keep it as debug-only using
snd_printdd(), and also add the rate-limit for avoiding floods.
Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=738681
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
No code altered at this point, simply preparing for upcoming
refactorizations.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi 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: Daniel Mack <zonque@gmail.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi 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: Daniel Mack <zonque@gmail.com>
Reported-by: Nicolai Krakowiak <nicolai.krakowiak@gmail.com>
Cc: stable@kernel.org
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi 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.conf
Signed-off-by: Guillaume Pellerin <yomguy@parisson.com>
Signed-off-by: Takashi 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: Clemens Ladisch <clemens@ladisch.de>
Cc: Daniel Mack <daniel@caiaq.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi 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: Felix Homann <fexpop@web.de>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi 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: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi 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: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stanse found that in snd_usb_parse_audio_endpoints, there is a
dangling pointer dereference. When snd_usb_parse_audio_format fails,
fp is freed, and continue invoked. On the next loop, there is
"fp && fp->altsetting == 1 && fp->channels == 1" test, but fp is set
from the last iteration (but is bogus) and thus ilegally dereferenced.
Set fp to NULL before "continue".
Signed-off-by: Jiri Slaby <jslaby@suse.cz>
Acked-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC2 devices have their information about pitch control stored in a
different field. Parse it, and emulate the bits for a v1 device.
A new struct uac2_iso_endpoint_descriptor is added.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds basic support for M-Audio's Fast Track Ultra series of USB
audio interfaces. It is a refactored version of the patch Clemens
Ladisch posted some time ago. Neither playback nor capturing work
properly at 44100 Hz (don't know why).
The other sampling rates work properly. There's no support for the DSP
mixer, yet.
Signed-off-by: Felix Homann <fexpop@web.de>
Signed-off-by: Takashi 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: Daniel Mack <daniel@caiaq.de>
Acked-by: Greg Kroah-Hartman <gregkh@suse.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sample rate setting is done with a 4-byte long class request that
addresses the interface.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi 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: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi 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: Daniel Mack <daniel@caiaq.de>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
