Takashi Iwai [Thu, 24 Mar 2011 08:50:15 +0000 (09:50 +0100)]
ALSA: Fix yet another race in disconnection
This patch fixes a race between snd_card_file_remove() and
snd_card_disconnect(). When the card is added to shutdown_files list
in snd_card_disconnect(), but it's freed in snd_card_file_remove() at
the same time, the shutdown_files list gets corrupted. The list member
must be freed in snd_card_file_remove() as well.
Reported-and-tested-by: Russ Dill <russ.dill@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 25 Mar 2011 02:25:48 +0000 (15:25 +1300)]
ALSA: asihpi - Update verbose debug print macros
Replace local VPRINTK1 with snd_printdd.
Create local snd_printddd instead of VPRINTK2 for most verbose debug.
In most cases let snd_printk supply default level for messages.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 25 Mar 2011 02:25:47 +0000 (15:25 +1300)]
ALSA: asihpi - Improve non-busmaster adapter operation
Make playback silence callback a no-op, card automatically outputs
silence when written data runs out.
Increasing update interval and thus minimum period avoids xrun on startup
or because of timer jitter.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Eliot Blennerhassett [Fri, 25 Mar 2011 02:25:46 +0000 (15:25 +1300)]
ALSA: asihpi - Support single-rate no-SRC cards
Cards without settable local samplerate and without SRC
still must have a valid samplerate.
This fixed rate is determined by reading the current rate for the card.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Mon, 7 Feb 2011 14:19:34 +0000 (15:19 +0100)]
ALSA: HDA: New
AD1984A model for Dell Precision R5500
For codec
AD1984A, add a new model to support Dell Precision R5500
or the microphone jack won't work correctly.
BugLink: http://bugs.launchpad.net/bugs/741516
Tested-by: Kent Baxley <kent.baxley@canonical.com>
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Benjamin Herrenschmidt [Fri, 25 Mar 2011 06:51:54 +0000 (17:51 +1100)]
ALSA: vmalloc buffers should use normal mmap
It's a big no-no to use pgprot_noncached() when mmap'ing such buffers
into userspace since they are mapped cachable in kernel space.
This can cause all sort of interesting things ranging from to garbled
sound to lockups on various architectures. I've observed that usb-audio
is broken on powerpc 4xx for example because of that.
Also remove the now unused snd_pcm_lib_mmap_noncached(). It's
an arch business to know when to use uncached mappings, there's
already hacks for MIPS inside snd_pcm_default_mmap() and other
archs are supposed to use dma_mmap_coherent().
(See my separate patch that adds dma_mmap_coherent() to powerpc)
Signed-off-by: Benjamin Herrenschmidt <benh@kernel.crashing.org>
CC: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 23 Mar 2011 21:54:32 +0000 (22:54 +0100)]
ALSA: hda - Fix SPDIF out regression on ALC889
The commit
5a8cfb4e8ae317d283f84122ed20faa069c5e0c4
ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization
changed to use the default initialization method for ALC889, but
this caused a regression on SPDIF output on some machines.
This seems due to the COEF setup included in the default init procedure.
For making SPDIF working again, the COEF-setup has to be avoided for
the id 0889.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=24342
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Keith A. Milner [Mon, 21 Mar 2011 20:15:08 +0000 (20:15 +0000)]
ALSA: usb-audio - Support for Boss JS-8 Jam Station
Signed-off-by: Keith A. Milner <maillist@superlative.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Thu, 17 Feb 2011 13:26:51 +0000 (14:26 +0100)]
ALSA: usb-audio: add Cakewalk UM-1G support
Add a quirk for the Cakewalk UM-1G USB MIDI interface in
"advanced driver" mode. (It already works in standard mode.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dan Rosenberg [Wed, 23 Mar 2011 15:42:57 +0000 (11:42 -0400)]
sound/oss/opl3: validate voice and channel indexes
User-controllable indexes for voice and channel values may cause reading
and writing beyond the bounds of their respective arrays, leading to
potentially exploitable memory corruption. Validate these indexes.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dan Rosenberg [Wed, 23 Mar 2011 14:53:41 +0000 (10:53 -0400)]
sound/oss: remove offset from load_patch callbacks
Was: [PATCH] sound/oss/midi_synth: prevent underflow, use of
uninitialized value, and signedness issue
The offset passed to midi_synth_load_patch() can be essentially
arbitrary. If it's greater than the header length, this will result in
a copy_from_user(dst, src, negative_val). While this will just return
-EFAULT on x86, on other architectures this may cause memory corruption.
Additionally, the length field of the sysex_info structure may not be
initialized prior to its use. Finally, a signed comparison may result
in an unintentionally large loop.
On suggestion by Takashi Iwai, version two removes the offset argument
from the load_patch callbacks entirely, which also resolves similar
issues in opl3. Compile tested only.
v3 adjusts comments and hopefully gets copy offsets right.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 23 Mar 2011 11:05:01 +0000 (12:05 +0100)]
Merge branch 'topic/asoc' into for-linus
David Henningsson [Wed, 23 Mar 2011 07:35:07 +0000 (08:35 +0100)]
ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
Similar to commit
7e59e097c09b82760bb0fe08b0fa2b704d76c3f4, this patch
avoids unnecessary volume control indices for more
Realtek auto-parsers, e g the ALC66x family, on the "Surround" and "Side"
controls.
These indices cause these volume controls to be ignored by PulseAudio and
vmaster and should be removed whenever possible.
Cc: stable@kernel.org
Reported-by: Jan Losinski <losinski@wh2.tu-dresden.de>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Tue, 22 Mar 2011 12:02:33 +0000 (12:02 +0000)]
ASoC: Support !REGULATOR build for sgtl5000
The regulator is optional depending on board design.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Lydia Wang [Tue, 22 Mar 2011 08:26:36 +0000 (16:26 +0800)]
ALSA: hda - VIA: Fix VT1708 can't build up Headphone control issue
Since VT1708 didn't support the control of getting connection number,
building of headphone control will fail in via_hp_build() function.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Tue, 22 Mar 2011 08:25:56 +0000 (16:25 +0800)]
ALSA: hda - VIA: Correct stream names for VT1818S
Correct stream names of analog playback and capture streams
for VT1818S.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Tue, 22 Mar 2011 08:25:03 +0000 (16:25 +0800)]
ALSA: hda - VIA: Fix codec type for VT1708BCE at the right timing
Add get_codec_type() in via_new_spec() function to make sure getting
correct codec type before building mixer controls.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Tue, 22 Mar 2011 08:24:10 +0000 (16:24 +0800)]
ALSA: hda - VIA: Fix invalid A-A path volume adjust issue
Modify vt_auto_create_analog_input_ctls() function to fix invalid a-a path
volume adjust issue for VT1708S, VT1702 and VT1716S codecs.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Tue, 22 Mar 2011 08:23:23 +0000 (16:23 +0800)]
ALSA: hda - VIA: Add missing support for VT1718S in A-A path
Modify mute_aa_path() function to support VT1718S codec.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Tue, 22 Mar 2011 08:22:37 +0000 (16:22 +0800)]
ALSA: hda - VIA: Fix independent headphone no sound issue
Modify via_independent_hp_put() function to support VT1718S and VT1812
codecs, and fix independent headphone no sound issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Tue, 22 Mar 2011 08:21:38 +0000 (16:21 +0800)]
ALSA: hda - VIA: Fix stereo mixer recording no sound issue
Modify function via_mux_enum_put() to fix stereo mixer recording
no sound issue.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Andres Mejia [Sun, 20 Mar 2011 18:07:40 +0000 (14:07 -0400)]
ALSA: hda - Set EAPD for Realtek ALC665
Set EAPD for Realtek ALC665 (Vendor Id: 0x10eSet EAPD for Realtek
ALC665 (Vendor Id: 0x10ec0665).
Signed-off-by: Andres Mejia <mcitadel@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 21 Mar 2011 11:00:00 +0000 (12:00 +0100)]
ALSA: usb - Remove trailing spaces from USB card name strings
Some USB devices give trailing spaces in strings returned from
usb_string(). This confuses the automatic card-id creation, resulting
always in "default".
This patch fixes the behavior by removing trailing spaces.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Xiaochen Wang [Fri, 18 Mar 2011 08:29:25 +0000 (16:29 +0800)]
sound: read i_size with i_size_read()
Convert direct read of inode->i_size to using i_size_read().
i_size_read is guaranteed to return a valid value and
its caller does not need to use addtional locking.
Signed-off-by: Xiaochen Wang <wangxiaochen0@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Wed, 16 Mar 2011 18:18:53 +0000 (18:18 +0000)]
ASoC: Remove bogus check for register validity in debugfs write
Since not all registers need to be cached and the cache is entirely
optional anyway we shouldn't be checking that a register is in the
cached range. If the register is invalid then the actual I/O code
can determine that and report an error.
Similarly, the step size can and should be enforced by the lower level
code if it's important.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Fri, 18 Mar 2011 10:51:13 +0000 (10:51 +0000)]
Merge branch 'topic/asoc' of git://git./linux/kernel/git/tiwai/sound-2.6 into for-2.6.39
Takashi Iwai [Fri, 18 Mar 2011 06:39:08 +0000 (07:39 +0100)]
Merge branch 'topic/misc' into for-linus
Dan Rosenberg [Thu, 17 Mar 2011 22:32:24 +0000 (18:32 -0400)]
ALSA: sound/pci/asihpi: check adapter index in hpi_ioctl
The user-supplied index into the adapters array needs to be checked, or
an out-of-bounds kernel pointer could be accessed and used, leading to
potentially exploitable memory corruption.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 18 Mar 2011 06:31:53 +0000 (07:31 +0100)]
ALSA: aloop - Fix possible IRQ lock inversion
loopback_pos_update() can be called in the timer callback, thus the lock
held should be irq-safe. Otherwise you'll get AB/BA deadlock together
with substream->self_group.lock.
Reported-and-tested-by: Knut Petersen <Knut_Petersen@t-online.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 16 Mar 2011 16:38:46 +0000 (17:38 +0100)]
Merge branch 'topic/hda' into for-linus
Takashi Iwai [Wed, 16 Mar 2011 16:38:41 +0000 (17:38 +0100)]
Merge branch 'topic/asoc' into for-linus
Nicolas Kaiser [Wed, 16 Mar 2011 16:10:11 +0000 (17:10 +0100)]
ALSA: sound/core: merge list_del()/list_add_tail() to list_move_tail()
Merge list_del() + list_add_tail() to list_move_tail().
Signed-off-by: Nicolas Kaiser <nikai@nikai.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Wed, 16 Mar 2011 11:14:46 +0000 (12:14 +0100)]
Merge branch 'for-2.6.39' of git://git./linux/kernel/git/lrg/asoc-2.6 into topic/asoc
Kirill A. Shutemov [Tue, 15 Mar 2011 22:53:24 +0000 (00:53 +0200)]
ALSA: ctxfi - use list_move() instead of list_del()/list_add() combination
Signed-off-by: Kirill A. Shutemov <kirill@shutemov.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Stephen Rothwell [Wed, 16 Mar 2011 06:39:28 +0000 (17:39 +1100)]
ALSA: firewire - msleep needs delay.h
fixes this error:
sound/firewire/fcp.c: In function 'fcp_avc_transaction':
sound/firewire/fcp.c:103: error: implicit declaration of function 'msleep'
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Tue, 15 Mar 2011 06:57:24 +0000 (07:57 +0100)]
ALSA: firewire-lib, firewire-speakers: handle packet queueing errors
Add an AMDTP stream error state that occurs when we fail to queue
another packet. In this case, the stream is stopped, and the error can
be reported when the application tries to restart the PCM stream.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Tue, 15 Mar 2011 06:55:50 +0000 (07:55 +0100)]
ALSA: firewire-lib: allocate DMA buffer separately
For correct cache coherency on some architectures, DMA buffers must be
allocated in a different cache line than data that is concurrently used
by the CPU.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Tue, 15 Mar 2011 06:55:02 +0000 (07:55 +0100)]
ALSA: firewire-lib: use no-info SYT for packets without SYT sample
In non-blocking mode, the SYT_INTERVAL is larger than the number of
audio frames in each packet, so there are packets that do not contain
any frame to which the SYT could be applied. For these packets, the
SYT must not be the timestamp of the next valid SYT frame, but the
special no-info SYT value.
This fixes broken playback on the FireWave at 44.1 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Tue, 15 Mar 2011 06:53:21 +0000 (07:53 +0100)]
ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver
Add a driver for two playback-only FireWire devices based on the OXFW970
chip.
v2: better AMDTP API abstraction; fix fw_unit leak; small fixes
v3: cache the iPCR value
v4: FireWave constraints; fix fw_device reference counting;
fix PCR caching; small changes and fixes
v5: volume/mute support; fix crashing due to pcm stop races
v6: fix build; one-channel volume for LaCie
v7: use signed values to make volume (range checks) work; fix function
block IDs for volume/mute; always use channel 0 for LaCie volume
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Acked-by: Stefan Richter <stefanr@s5r6.in-berlin.de>
Tested-by: Jay Fenlason <fenlason@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 14 Mar 2011 14:53:15 +0000 (15:53 +0100)]
ALSA: hda - Remove an unused variable in patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vitaliy Kulikov [Thu, 10 Mar 2011 19:43:35 +0000 (13:43 -0600)]
ALSA: hda - pin-adc-mux-dmic auto-configuration of 92HD8X codecs
This patch replaces use of the harcoded arrays of pins, muxes, digital
mics and adcs with the auto-generated ones using codec parsing and
auto-discovers all actually connected digital mic pins on 92HD8X-like
codecs
This patch also adds the support for d-mic on pin 0x20.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Vitaliy Kulikov [Thu, 10 Mar 2011 01:47:43 +0000 (19:47 -0600)]
ALSA: hda - fix digital mic selection in mixer on 92HD8X codecs
When the mux for digital mic is different from the mux for other mics,
the current auto-parser doesn't handle them in a right way but provides
only one mic. This patch fixes the issue.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 10 Mar 2011 13:11:59 +0000 (14:11 +0100)]
ALSA: hda - Move default input-src selection to init part
Move the default input-src selection code for alc268/269 to the init
part instead of the parser. The input-src selection might be overwritten
by init verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Thu, 10 Mar 2011 11:51:11 +0000 (12:51 +0100)]
ALSA: hda - Initialize special cases for input src in init phase
Currently some special handling for the unusual case like dual-ADCs
or a single-input-src is done in the tree-parse time in
set_capture_mixer(). But this setup could be overwritten by static
init verbs.
This patch moves the initialization into the init phase so that
such input-src setup won't be lost.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Przemyslaw Bruski [Sun, 13 Mar 2011 15:18:58 +0000 (16:18 +0100)]
ALSA: ctxfi - Clear input settings before initialization
Clear input settings before initialization.
Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Przemyslaw Bruski [Sun, 13 Mar 2011 15:18:57 +0000 (16:18 +0100)]
ALSA: ctxfi - Fix SPDIF status retrieval
SDPIF status retrieval always returned the default settings instead of
the actual ones.
Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Przemyslaw Bruski [Sun, 13 Mar 2011 15:18:56 +0000 (16:18 +0100)]
ALSA: ctxfi - Fix incorrect SPDIF status bit mask
SPDIF status mask creation was incorrect.
Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Przemyslaw Bruski [Sun, 13 Mar 2011 15:18:55 +0000 (16:18 +0100)]
ALSA: ctxfi - Fix microphone boost codes/comments
microphone boost was set at +12dB, not +20dB (like in Windows driver
and in adc_conf structure declaration), some comments added.
Signed-off-by: Przemyslaw Bruski <pbruskispam@op.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 11 Mar 2011 14:45:32 +0000 (15:45 +0100)]
ALSA: atiixp - Fix wrong time-out checks during ac-link reset
The time-out in snd_atiixp_aclink_reset() is wrongly checked, and
it resulted in exiting from the loop at the first iteration.
Reported-by: Amir Shamsuddin <AmirS2+alsa@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Paul Bolle [Fri, 11 Mar 2011 10:41:06 +0000 (11:41 +0100)]
ALSA: intel8x0m: append 'm' to "r_intel8x0"
Appending an 'm' will distinguish it from a similar struct in intel8x0.c
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Paul Bolle [Fri, 11 Mar 2011 10:37:55 +0000 (11:37 +0100)]
ALSA: intel8x0m: add 'm' as "suffix" to static functions
Adding an 'm' will distinguish them from identical names in intel8x0.c.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Paul Bolle [Fri, 11 Mar 2011 10:24:52 +0000 (11:24 +0100)]
ALSA: intel8x0m: wait a bit before warm reset check
At every resume a laptop I use prints this message (at KERN_ERR level):
ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2]
The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what
seems to be happening is that the register involved indicated a warm
reset for some time (as the ICH_AC97WARM bit was set) but by the time
the warning is printed, and that same register is checked again, that
bit is already cleared and only the ICH_AC97COLD bit is still set.
It turns out a warm reset needs some time to settle, but it is currently
checked right away. The test therefore fails the first time it is done
and schedule_timeout_uninterruptible() will be called. Once we return
from that jiffies is already (far) past end_time on this laptop, so we
exit the loop, print a warning, and exit the function while the warm
reset actually succeeded.
A way to fix this is to call usleep_range() after writing to the
register involved. A handful of tests suggest 500 usecs is a safe value.
(This might punish the "finish cold reset" case, but on this laptop such
a cold reset apparently never happens, so I can't say for sure.)
While we're at it drop the extra single tick from end_time, as it looks
rather silly.
Signed-off-by: Paul Bolle <pebolle@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Oliver Neukum [Fri, 11 Mar 2011 13:51:12 +0000 (14:51 +0100)]
ALSA: usbaudio: implement USB autosuspend
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: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Oliver Neukum [Fri, 11 Mar 2011 12:19:43 +0000 (13:19 +0100)]
ALSA: usbaudio: fix suspend/resume
- ESHUTDOWN must be correctly handled
- the optional interrupt endpoint's URB must be stopped and restarted
Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 11 Mar 2011 13:48:09 +0000 (14:48 +0100)]
Merge branch 'fix/misc' into topic/misc
Marek Belisko [Wed, 9 Mar 2011 20:46:20 +0000 (21:46 +0100)]
ASoC: mini2440: Fix uda134x codec problem.
ASoC audio for mini2440 platform in current kenrel doesn't work.
First problem is samsung_asoc_dma device is missing in initialization.
Next problem is with codec. Codec is initialized but never probed
because no platform_device exist for codec driver. It leads to errors
during codec binding to asoc dai. Next problem was platform data which
was passed from board to asoc main driver but not passed to codec when
called codec_soc_probe().
Following patch should fix issues. But not sure if in correct way.
Please review.
Signed-off-by: Marek Belisko <marek.belisko@open-nandra.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Mark Brown [Fri, 11 Mar 2011 12:07:31 +0000 (12:07 +0000)]
ASoC: Fix spacing in MAX8950
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Vasily Khoruzhick [Fri, 11 Mar 2011 09:05:38 +0000 (11:05 +0200)]
ASoC: PXA: Z2: Fix codec pin name
MONO was renamed to MONO1.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
Vasily Khoruzhick [Fri, 11 Mar 2011 09:05:37 +0000 (11:05 +0200)]
ASoC: PXA: z2: Mute internal speaker when headphones are connected
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Christian Glindkamp [Wed, 9 Mar 2011 10:20:04 +0000 (11:20 +0100)]
ASoC: Add MAX9850 codec driver
This patch adds ASoC support for the MAX9850 codec with headphone
amplifier.
Supported features:
- Playback
- 16, 20 and 24 bit audio
- 8k - 48k sample rates
- DAPM
Signed-off-by: Christian Glindkamp <christian.glindkamp@taskit.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Fri, 29 Aug 2008 14:09:01 +0000 (16:09 +0200)]
ALSA: Add snd_ctl_activate_id()
Added a new API function snd_ctl_activate_id() for activate / inactivate
the control element dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Adam Lackorzynski [Thu, 10 Mar 2011 16:41:56 +0000 (17:41 +0100)]
ALSA: hda: Prevent writing ICH6_PCIREG_TCSEL on AMD systems
azx_init_pci() always writes PCI config register ICH6_PCIREG_TCSEL
although this looks to be only defined on Intel systems and has a
different meaning on AMD systems. On AMD systems the PCI interrupt pin
control register is modified instead.
Since the meaning of offset 0x44 in device specific configuration space is
unknown for devices by other vendors, we only exclude AMD systems to
retain the current behaviour.
Signed-off-by: Adam Lackorzynski <adam@os.inf.tu-dresden.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Fri, 4 Mar 2011 15:54:52 +0000 (16:54 +0100)]
ALSA: HDA: Realtek ALC88x: Do not over-initialize speakers and hp that are primary outputs
Do not initialize again the what has already been initialized as
multi outs, as this breaks surround speakers.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Fri, 4 Mar 2011 13:22:25 +0000 (14:22 +0100)]
ALSA: HDA: Fixup unnecessary volume control index on Realtek ALC88x
Without this change, a volume control named "Surround" or "Side" would
get an unnecessary index, causing it to be ignored by the vmaster and
PulseAudio.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Fri, 4 Mar 2011 13:08:30 +0000 (14:08 +0100)]
ALSA: HDA: Fix volume control naming for surround speakers on Realtek auto-parser
When more than one pair of internal speakers is present, allow names
according to their channels.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
David Henningsson [Fri, 4 Mar 2011 12:37:50 +0000 (13:37 +0100)]
ALSA: HDA: Enable surround and subwoofer on Lenovo Ideapad Y530
The pin config values would change the association instead of the
sequence, this commit fixes that up.
Tested-by: Bartłomiej Żogała <nusch88@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Alexey Fisher [Thu, 10 Mar 2011 13:53:39 +0000 (14:53 +0100)]
ALSA: usb-audio - Add "cval->res = 384" quirk for Logitech Webcam C600
One more affected devices: Logitech Webcam C600 (046d:0808)
Volume range before quirk is 6400, after (also real) is 16.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Alexey Fisher [Thu, 10 Mar 2011 13:53:38 +0000 (14:53 +0100)]
ALSA: usb-audio - Add volume range check and warn if it too big
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Thu, 10 Mar 2011 14:16:35 +0000 (14:16 +0000)]
Merge branch 'for-2.6.38' into for-2.6.39
Mark Brown [Wed, 9 Mar 2011 19:10:15 +0000 (19:10 +0000)]
ASoC: Ensure WM8958 gets all WM8994 late revision widgets
Without this fix the driver won't instantiate properly on relevant
devices.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Mark Brown [Wed, 9 Mar 2011 19:09:17 +0000 (19:09 +0000)]
ASoC: Fix typo in late revision WM8994 DAC2R name
Without this fix the driver won't instantiate properly on relevant
devices.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
Takashi Iwai [Wed, 9 Mar 2011 16:33:48 +0000 (17:33 +0100)]
sound: Use sound_register_*() for additional OSS minor devices
Since OSS driver creates the device entries for /dev/audio* and
/dev/dspW* by itself without coping with sound_core, it leads to
conflicts with others and let sysfs spewing warnings.
This patch rewrites the registration part of OSS driver to use
the standard method also for additional minor devices.
Reported-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl)
Tested-by: Steven Rostedt <rostedt@goodmis.org> (with ktest.pl)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Alexander Sverdlin [Mon, 7 Mar 2011 17:30:12 +0000 (20:30 +0300)]
ASoC: Enable 192kHz sample rate for EP93xx.
Enable 192kHz sample rate for EP93xx.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Alexander Sverdlin [Mon, 7 Mar 2011 17:30:36 +0000 (20:30 +0300)]
ASoC: Improve EP93xx I2S clocks management.
Improve EP93xx I2S clocks management.
Some freqs values are set not exact as they requested for MCLK and
original code was not able to find divisors for SCLK and LRCLK.
This code just picks up nearest value from 3 possible variants.
This patch makes 44100 and 192000 rates working and fixes
capture function (by selecting SCLK/LRCLK=64 where possible).
All other rates should work as before.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Alexander Sverdlin [Mon, 7 Mar 2011 17:29:53 +0000 (20:29 +0300)]
ASoC: EDB93xx: Manage I2S rates according to datasheet for CS4271 CODEC.
Manage I2S rates according to datasheet for CS4271 CODEC in EDB93xx
machine driver.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Alexander Sverdlin [Mon, 7 Mar 2011 17:29:45 +0000 (20:29 +0300)]
ASoC: Manage mode and rate bits correctly for CS4271 CODEC.
Manage mode and rate bits correctly, according to datasheet in CS4271 CODEC.
This is done to make capture work properly.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 9 Mar 2011 12:37:42 +0000 (12:37 +0000)]
Merge branch 'for-2.6.38' into for-2.6.39
Conflicts:
sound/soc/codecs/wm8978.c
sound/soc/soc-dapm.c
Mark Brown [Wed, 9 Mar 2011 11:33:09 +0000 (11:33 +0000)]
ASoC: Fix double addition of prefixes due to widget prefixing
We're not only prefixing all controls, we're also prefixing the widget
names in the runtime data. This causes us to add the prefix twice - once
when using the widget name to generate the control name and once when
adding the control.
Really we shouldn't be prefixing the widget names at all, the matching
code should be handing this as we always know which DAPM context a
widget came from and always display the widget name in terms of a DAPM
context. However, we're quite close to the merge window and that's
relatively invasive.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Reported-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 8 Mar 2011 19:29:53 +0000 (19:29 +0000)]
ASoC: Use the correct DAPM context when cleaning up final widget set
Now we've got multi-component we need to make sure that the DAPM context
(and hence register I/O context) we use to apply the pending updates at
the end of a DAPM sequence is the one we were processing rather than the
one that was used to initate the state change.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Wed, 2 Mar 2011 11:01:18 +0000 (11:01 +0000)]
ASoC: Fix broken bitfield definitions in WM8978
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
Mark Brown [Tue, 8 Mar 2011 18:52:08 +0000 (18:52 +0000)]
ASoC: Simplify WM9081 speaker startup by using widgets for spaker output
Now we have a register write minimisation code in DAPM we don't need to
worry about the ordering of the enable and disable of the PGA and the
output stage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Jarkko Nikula [Wed, 9 Mar 2011 09:25:00 +0000 (11:25 +0200)]
ASoC: omap: rx51: Enable McBSP2 sidetone
McBSP sidetone is needed in telephony applications. McBSP sidetone is a
configurable FIR filter that forms a loopback from McBSP input to output.
This patch enables the McBSP2 sidetone ALSA controls so that it can be used
on Nokia RX-51/N900.
Sidetone feature can be tested with following commands:
(set up codec input and output paths)
# Enable and configure sidetone
amixer -D hw:0 set 'McBSP2 Sidetone' on
amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767
echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps
# Do not loop audio via CPU
arecord -f dat >/dev/null |aplay /dev/zero
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Dan Carpenter [Tue, 8 Mar 2011 11:39:24 +0000 (14:39 +0300)]
ASoC: sgtl5000: use after free in ldo_regulator_register()
The "ldo" variable was dereferenced after free on the error path.
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 8 Mar 2011 17:23:24 +0000 (17:23 +0000)]
ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew()
Currently will ignore prefixes when creating DAPM controls. Since currently
all control creation goes through snd_soc_cnew() we can fix this by factoring
the prefixing into that function.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 8 Mar 2011 00:17:56 +0000 (00:17 +0000)]
ASoC: Warn rather than set a silly constraint when we can't do symmetry
Symmetric rate configuration can fail if the second stream starting tries
to apply the symmetric constraint before the first stream has got far
enough to pick a rate. Rather than try to enforce a nonsensical rate of
0Hz log a warning and allow the application to carry on. Things might go
wrong later on but the user will know about it and there's unlikely to be
lasting damage.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Tue, 8 Mar 2011 00:11:15 +0000 (00:11 +0000)]
ASoC: Refactor symmetric_rates check to reduce indentation
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 7 Mar 2011 20:59:45 +0000 (20:59 +0000)]
ASoC: Convert WM9081 SYSCLK configuration to be device wide
Also respace the CODEC ops a bit for legibility.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 7 Mar 2011 20:58:11 +0000 (20:58 +0000)]
ASoC: Provide CODEC clocking operations and API calls
When multi component systems use DAIless amplifiers which require clocking
configuration it is at best hard to use the current clocking API as this
requires a DAI even though the device may not even have one. Address this
by adding set_sysclk() and set_pll() operations and APIs for CODECs.
In order to avoid issues with devices which could be used either with or
without DAIs make the DAI variants call through to their CODEC counterparts
if there is no DAI specific operation. Converting over entirely would create
problems for multi-DAI devices which offer per-DAI clocking setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 7 Mar 2011 16:42:20 +0000 (16:42 +0000)]
ASoC: Fix section mismatch warnings in WM8994
Annoying as the __devinitdata is actually correct.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 7 Mar 2011 16:39:50 +0000 (16:39 +0000)]
ASoC: Use data based init for WM9081 DAPM
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Mon, 7 Mar 2011 16:38:44 +0000 (16:38 +0000)]
ASoC: Add DAPM widget and path data to CODEC driver structure
Allow a slight simplification of CODEC drivers by allowing DAPM routes and
widgets to be provided in a table. They will be instantiated at the end of
CODEC probe.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Takashi Iwai [Tue, 8 Mar 2011 17:20:46 +0000 (18:20 +0100)]
ALSA: asihpi - Use %zd for size_t argument in error message
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 7 Mar 2011 12:24:30 +0000 (13:24 +0100)]
ALSA: control: clean up snd_ctl_hole_check()
The return value of snd_ctl_hole_check() is used only to detect whether
to continue the loop in snd_ctl_find_hole() or not, so we can simplify
the code by changing this return type to a boolean. Also rename this
function to better show what it actually does.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 7 Mar 2011 12:22:50 +0000 (13:22 +0100)]
ALSA: control: fix numid conflict check for new controls
The purpose of the snd_ctl_hole_check() function is to find conflicts
between the numerical IDs of the new control and those of any existing
controls. However, it would fail to detect an existing control whose
count is smaller than the new control's count and whose interval of IDs
is entirely contained in the interval of the new control's IDs.
To fix this, use the correct formula to detect overlapping intervals,
which happens to simplify the condition.
This problem was not encountered so far because ALSA does not yet allow
drivers to allocate specific control IDs.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Adrian Knoth [Mon, 7 Mar 2011 18:10:11 +0000 (19:10 +0100)]
ALSA: hdspm - Add firmware ID for older AES32 cards.
The current AES32 firmware revision ID is 234, however, a user confirmed
that everything works fine with the previous revision, too.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Linus Torvalds [Tue, 8 Mar 2011 05:09:37 +0000 (21:09 -0800)]
Linux 2.6.38-rc8
Linus Torvalds [Tue, 8 Mar 2011 04:46:39 +0000 (20:46 -0800)]
Merge branch 's5p-fixes-for-linus' of git://git./linux/kernel/git/kgene/linux-samsung
* 's5p-fixes-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/kgene/linux-samsung:
ARM: S3C64XX: Update regulator names for debugfs compatiblity on SMDK6410
ARM: S3C64XX: Fix build with WM1190 disabled and WM1192 enabled on SMDK6410
ARM: S3C64XX: Reduce output of s3c64xx_dma_init1()
ARM: S3C64XX: Tone down SDHCI debugging
ARM: S3C64XX: Add clock for i2c1
ARM: S3C64XX: Staticise non-exported GPIO to interrupt functions
ARM: SAMSUNG: Include devs.h in dev-uart.c to prototype devices
ARM: S3C64XX: Fix keypad setup to configure correct number of rows
ARM: S3C2440: Fix usage gpio bank j pin definitions on GTA02
ARM: S5P64X0: Fix number of GPIO lines in Bank F
ARM: S3C2440: Select missing S3C_DEV_USB_HOST on GTA02
Linus Torvalds [Tue, 8 Mar 2011 04:45:42 +0000 (20:45 -0800)]
Merge branch 'fixes' of /home/rmk/linux-2.6-arm
* 'fixes' of master.kernel.org:/home/rmk/linux-2.6-arm:
davinci: cpufreq: fix section mismatch warning
DaVinci: fix compilation warnings in <mach/clkdev.h>
davinci: tnetv107x: fix register indexing for GPIOs numbers > 31
davinci: da8xx/omap-l1x: add platform device for davinci-pcm-audio
ARM: pxa/tosa: register wm9712 codec device
ARM: pxa: enable pxa-pcm-audio on pxa210/pxa25x platform
ARM: pxa/colibri: don't register pxa2xx-pcmcia nodes on non-colibri platforms
ARM: pxa/tosa: drop setting LED trigger name, as it's unsupported now
ARM: 6762/1: Update number of VIC for S5P6442 and S5PC100
ARM: 6761/1: Update number of VIC for S5PV210
ARM: 6768/1: hw_breakpoint: ensure debug logic is powered up on v7 cores
ARM: 6767/1: ptrace: fix register indexing in GETHBPREGS request
ARM: 6765/1: remove obsolete comment from asm/mach/arch.h
ARM: 6757/1: fix tlb.h induced linux/swap.h build failure
Linus Torvalds [Tue, 8 Mar 2011 04:45:12 +0000 (20:45 -0800)]
Merge branch 'for-linus' of git://git./linux/kernel/git/cjb/mmc
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/cjb/mmc:
mmc: sdio: Allow sdio operations in other threads during sdio_add_func()
Linus Torvalds [Tue, 8 Mar 2011 04:43:55 +0000 (20:43 -0800)]
Merge branch 'drm-fixes' of git://git./linux/kernel/git/airlied/drm-2.6
* 'drm-fixes' of git://git.kernel.org/pub/scm/linux/kernel/git/airlied/drm-2.6:
drm: index i shadowed in 2nd loop
drm/nv50-nvc0: prevent multiple vm/bar flushes occuring simultanenously
drm/nouveau: fix regression causing ttm to not be able to evict vram
drm/i915: Rebind the buffer if its alignment constraints changes with tiling
drm/i915: Disable GPU semaphores by default
drm/i915: Do not overflow the MMADDR write FIFO
Revert "drm/i915: fix corruptions on i8xx due to relaxed fencing"
projects / GitHub / LineageOS / android_kernel_motorola_exynos9610.git / log