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This moves the lock_kernel() call from soundcore_open
to the individual OSS device drivers, where we can deal
with it one driver at a time if needed, or just kill
off the drivers.
All core components in ALSA already provide
adequate locking in their open()-functions
and do not require the big kernel lock, so
there is no need to add the BKL there.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As per-stream volume controls, the DXS controls are not intended to
adjust the overall sound level and so are initialized every time
a stream is opened. However, there are special situations where one
wants to reduce the overall volume in the digital domain, i.e., before
the AC'97 codec's PCM volume control. To allow this, add a module
parameter that sets the initial DXS volume.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Soeren D. Schulze <soeren.d.schulze@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It is not advisable to print a warning when a device does not support
setting the sample rate because this is perfectly valid for devices with
a single rate or where rates are implicitly changed by selecting another
alternate setting.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Handle errors in tuner level caching,
Ccorrect error code for aesebu rx status.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Compander API changed to one function per parameter.
Factor out some common code for stereo log value reading.
Make some more entity functions static.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Remove some deprecated items.
Change compander api to one function per parameter.
Add a version string define.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Most of this function is protected by the sound_loader_lock.
We can push down the BKL to this call out err = file->f_op->open(inode,file);
In order to build the sound core without the BKL, we
will need to push the lock_kernel() call into the ~20
device drivers that register their file operations.
Signed-off-by: John Kacur <jkacur@redhat.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Alan Cox <alan@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A few boards using this controller are reported to need a little extra
time during their reset cycle.
Reported-by: Michael Goeke <michael.goeke@icachip.de>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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When using a timing voice to clock out periods during capture, the
driver would slowly loose synchronization and never catch up, eventually
reaching a point where it no longer generated interrupts. To avoid
this situation, the virtual period clocking was changed to shorten the
next timing period when our timing voice falls too far behind the
capture voice. In addition, the first virtual period for the timing
voice was slightly too short, causing the timing voice to initially be
ahead of the capture voice.
While tracking down this problem, I noticed that the expected sample
offset was being incorrectly initialized, causing an overrun to be
incorrectly reported when the timing voice happened to be perfectly
synchronized.
Reported-by: Hans Schou <linux@schou.dk>
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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When using poll() to wait for the next period -- or avail_min samples --
one gets a consistent delay for each system call that is usually just a
little short of the selected period time. However, When using
snd_pcm_read/write(), one gets a jittery delay that alternates between
less than a millisecond and approximately two period times. This is
caused by snd_pcm_lib_{read,write}1() transferring any available samples
to the user's buffer and adjusting the application pointer prior to
sleeping to the end of the current period. When the next period
interrupt occurs, there is then less than avail_min samples remaining to
be transferred in the period, so we end up sleeping until a second
period occurs.
This is solved by using runtime->twake as the number of samples needed
for a wakeup in addition to selecting the proper wait queue to wake in
snd_pcm_update_state(). This requires twake to be non-zero when used
by snd_pcm_lib_{read,write}1() even if avail_min is zero.
Signed-off-by: Dave Dillow <dave@thedillows.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Replaced the forgotten cval->mixer->ctrlif.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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Also add a list of open topics.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some programs like Skype trying to set capture volume automatically.
Normally it will tray, carefully step by step lover or higher, set the volume.
In real word it work not really well, because devises and vendors lie about
real audio settings.
For example most Logitech webcams have 6400 or 3500 steps for capture volume.
They do not tell that actual resolution is 384. So we have only 7 or 18 real
steps. In this patch I set real resolution only for tested devices.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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Signed-off-by: Andy Shevchenko <ext-andriy.shevchenko@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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These give incorrect results for index wrap on 64 bit.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For RANGE requests, we should only query as much bytes as we're in fact
interested in.
For CUR requests, we shouldn't confuse the firmware with an overlong
request but just ask for 2 bytes.
This might need fixing in the future as it's not entirely clear when to
dispatch 1-byte, 2-byte and 4-byte request blocks. For now, we assume
everything is coded in 16bit - this works for all firmware
implementations I've seen.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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A device may report its supported sample rates in ranges rather than in
discrete triplets. The code used to only parse the MIN field instead of
properly paying attention to the MAX and RES values.
Also, handle RES values of 1 correctly and announce a continous sample
rate range in this case.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Control messages directed to an interface must have the interface number
set in the lower 8 bits of wIndex. This wasn't done correctly for some
clock and mixer messages.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Reported-by: Alex Lee <alexlee188@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The '*bitclk' of structure 'snd_at73c213' seems no use,
so I make a patch to remove the unnecessary variable.
Signed-off-by: Wan ZongShun <mcuos.com@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda-intel - fix wallclk variable update and condition
ALSA: asihpi - Fix uninitialized variable
ALSA: hda: Use LPIB for ASUS M2V
usb/gadget: Replace the old USB audio FU definitions in f_audio.c
ASoC: MX31ads sound support should depend on MACH_MX31ADS_WM1133_EV1
ASoC: Add missing Kconfig entry for Phytec boards
ALSA: usb-audio: export UAC2 clock selectors as mixer controls
ALSA: usb-audio: clean up find_audio_control_unit()
ALSA: usb-audio: add UAC2 sepecific Feature Unit controls
ALSA: usb-audio: unify constants from specification
ALSA: usb-audio: parse clock topology of UAC2 devices
ALSA: usb-audio: fix selector unit string index accessor
include/linux/usb/audio-v2.h: add more UAC2 details
ALSA: usb-audio: support partially write-protected UAC2 controls
ALSA: usb-audio: UAC2: clean up parsing of bmaControls
ALSA: hda: Use LPIB for another mainboard
ALSA: hda: Use mb31 quirk for an iMac model
ALSA: hda: Use LPIB for an ASUS device
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This patch fixes thinko introduced in "last minutes" before commiting
of the last wallclk patch.
It also fixes the condition checking if the first period after last
wallclk update is processed. There is a little rounding error in
period_wallclk.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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This patch fixes thinko introduced in "last minutes" before commiting
of the last wallclk patch.
It also fixes the condition checking if the first period after last
wallclk update is processed. There is a little rounding error in
period_wallclk.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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Initialize prev_ctl properly before reference:
sound/pci/asihpi/asihpi.c: In function ‘snd_card_asihpi_mixer_new’:
sound/pci/asihpi/asihpi.c:2568:30: warning: ‘prev_ctl.dst_node_index’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Grant patches added an of mach table to struct device_driver. However,
while he changed the macio device code to use that, he left the match
table pointer in struct macio_driver and didn't update drivers to use
the "new" one, thus breaking the probing.
This completes the change by moving all drivers to setup the "new"
one, removing all traces of the old one, and while at it (since it
changes the exact same locations), I also remove two other duplicates
from struct driver which are the name and owner fields.
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
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BugLink: https://launchpad.net/bugs/587546
Symptom: On the reporter's ASUS M2V, using PulseAudio in Ubuntu 10.04 LTS
results in the PA daemon crashing shortly after attempting playback of an
audio file.
Test case: Using Ubuntu 10.04 LTS (Linux 2.6.32.12), Linux 2.6.33, or
Linux 2.6.34, attempt playback of an audio file while PulseAudio is
active.
Resolution: add SSID for this machine to the position_fix quirk table,
explicitly specifying the LPIB method.
Reported-and-Tested-By: D Tangman
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The UAC2 clock selectors are fortunately compatible with UAC1 audio
selector units, so we can simply reuse the same approach to get all the
linked units.
Requests to this control need a different CS value though.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use a struct to parse the audio units, and return usable descriptors
for all types. There's no need to limit the result set, except for some
kind of sanity check.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The bits to enable them are always 0 for UAC1 devices, so no additional
checks are required.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Move more definitions from private enums to appropriate header files.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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So far, UAC2 controls are marked read-only if any of the channels are
marked read-only in the descriptors. Change this behaviour and
- mark them writeable unless all channels are read-only
- store the read-only mask in usb_mixer_elem_info and
- check the mask again in set_cur_mix_value(), and bail out for
write-protected channels.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Arnd Bergmann
Clemens Ladisch
Eliot Blennerhassett
John Kacur
Takashi Iwai
David Dillow
Daniel Mack
Alexey Fisher
Jiri Slaby
Andy Shevchenko
Wan ZongShun
Yegor Yefremov
Linus Torvalds
Jaroslav Kysela
Benjamin Herrenschmidt
Daniel T Chen
Sascha Hauer