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Commit Graph

234338 Commits

Author SHA1 Message Date
Takashi Iwai
ce24f58a11 Merge branch 'topic/asoc' into for-linus 2011-03-23 12:05:01 +01:00
David Henningsson
5a8826463c ALSA: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
Similar to commit 7e59e097c0, 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>
2011-03-23 09:22:02 +01:00
Mark Brown
333802e90d 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>
2011-03-22 18:26:30 +00:00
Lydia Wang
ee3c35c082 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>
2011-03-22 12:56:06 +01:00
Lydia Wang
970f630f5a 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>
2011-03-22 12:56:01 +01:00
Lydia Wang
0341ccd755 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>
2011-03-22 12:54:32 +01:00
Lydia Wang
169222813e 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>
2011-03-22 12:54:14 +01:00
Lydia Wang
ab657e0cac 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>
2011-03-22 12:53:52 +01:00
Lydia Wang
ce0e5a9e81 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>
2011-03-22 12:42:56 +01:00
Lydia Wang
bff5fbf50b 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>
2011-03-22 12:42:23 +01:00
Andres Mejia
75eb1c311d 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>
2011-03-21 12:32:50 +01:00
Takashi Iwai
3ffc1222bd 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>
2011-03-21 12:32:44 +01:00
Xiaochen Wang
977a6ef3c0 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>
2011-03-18 15:14:57 +01:00
Mark Brown
b1a56b331a 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>
2011-03-18 10:51:42 +00:00
Mark Brown
2a3887f701 Merge branch 'topic/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.39 2011-03-18 10:51:13 +00:00
Takashi Iwai
d351cf4603 Merge branch 'topic/misc' into for-linus 2011-03-18 07:39:08 +01:00
Dan Rosenberg
4a122c10fb 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>
2011-03-18 07:38:29 +01:00
Takashi Iwai
98d21df431 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>
2011-03-18 07:31:53 +01:00
Takashi Iwai
433e8327ca Merge branch 'topic/hda' into for-linus 2011-03-16 17:38:46 +01:00
Takashi Iwai
27b92d4ff2 Merge branch 'topic/asoc' into for-linus 2011-03-16 17:38:41 +01:00
Nicolas Kaiser
5b7c757d1a 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>
2011-03-16 17:35:16 +01:00
Takashi Iwai
e58a8947b0 Merge branch 'for-2.6.39' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc 2011-03-16 12:14:46 +01:00
Kirill A. Shutemov
9d4ed9e077 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>
2011-03-16 07:48:40 +01:00
Stephen Rothwell
5de0ee574b 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>
2011-03-16 07:47:33 +01:00
Clemens Ladisch
ec00f5e444 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>
2011-03-15 08:42:30 +01:00
Clemens Ladisch
5b2599a07e 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>
2011-03-15 08:42:28 +01:00
Clemens Ladisch
be45436632 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>
2011-03-15 08:42:25 +01:00
Clemens Ladisch
31ef9134eb 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>
2011-03-15 08:42:22 +01:00
Takashi Iwai
cc90fd725e ALSA: hda - Remove an unused variable in patch_realtek.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-03-14 15:53:15 +01:00
Vitaliy Kulikov
699d899560 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>
2011-03-14 15:38:57 +01:00
Vitaliy Kulikov
094a42452a 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>
2011-03-14 15:21:17 +01:00
Takashi Iwai
ae0ebbf70a 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>
2011-03-14 15:02:37 +01:00
Takashi Iwai
584c0c4c35 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>
2011-03-14 15:02:14 +01:00
Przemyslaw Bruski
efed5f2666 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>
2011-03-14 11:19:43 +01:00
Przemyslaw Bruski
f164753a26 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>
2011-03-14 11:19:42 +01:00
Przemyslaw Bruski
4c1847e884 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>
2011-03-14 11:19:30 +01:00
Przemyslaw Bruski
19002fd5f1 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>
2011-03-14 11:15:59 +01:00
Takashi Iwai
7e79443ce2 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>
2011-03-11 15:45:32 +01:00
Paul Bolle
966a7f0dc4 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>
2011-03-11 15:22:05 +01:00
Paul Bolle
a6e8509f21 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>
2011-03-11 15:22:02 +01:00
Paul Bolle
5cd2ad81f9 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>
2011-03-11 15:22:00 +01:00
Oliver Neukum
88a8516a21 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>
2011-03-11 14:59:29 +01:00
Oliver Neukum
edf7de31c2 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>
2011-03-11 14:51:51 +01:00
Takashi Iwai
cc99a0861f Merge branch 'fix/misc' into topic/misc 2011-03-11 14:48:09 +01:00
Marek Belisko
a110f4ef81 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
2011-03-11 12:17:11 +00:00
Mark Brown
27380fb830 ASoC: Fix spacing in MAX8950
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-11 12:07:31 +00:00
Vasily Khoruzhick
64c25a92e8 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
2011-03-11 12:03:13 +00:00
Vasily Khoruzhick
5f3822c48a 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>
2011-03-11 12:03:03 +00:00
Christian Glindkamp
0e45cab644 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>
2011-03-11 12:01:44 +00:00
Takashi Iwai
3cbdd75331 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>
2011-03-11 10:49:15 +00:00