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commit 030b655b062fe5190fc490e0091ea50307d7a86f upstream
Correct headphone detection for 1st generation iMac G3 Slot-loading (Screamer).
This patch fixes the regression in the recent snd-powermac which
doesn't support some G3/G4 PowerMacs:
http://lkml.org/lkml/2008/10/1/220
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Tested-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit 4dbf95ba6c344186ec6d38ff514dc675da464bec upstream
Add mixer controls for PowerMac G4 AGP (Screamer).
This patch fixes the regression in the recent snd-powermac which
doesn't support some G3/G4 PowerMacs:
http://lkml.org/lkml/2008/10/1/220
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Tested-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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Upstream-commit-id: 24e8fc498e9618338854bfbcf8d1d737e0bf1775
The power_mutex lock in snd_pcm_drop may cause a possible deadlock
chain, and above all, it's unneeded. Let's get rid of it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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Upstream-commit-id: 399ccdc1cd4e92e541d4dacbbf18c52bd693418b
The PCM and rawmidi open callbacks have a lock against card->controls_list
but it takes a wrong one, card->controls_rwsem, instead of a right one
card->ctl_files_rwlock. This patch fixes them.
This change also fixes automatically the potential deadlocks due to
mm->mmap_sem in munmap and copy_from/to_user, reported by Sitsofe
Wheeler:
A: snd_ctl_elem_user_tlv(): card->controls_rwsem => mm->mmap_sem
B: snd_pcm_open(): card->open_mutex => card->controls_rwsem
C: munmap: mm->mmap_sem => snd_pcm_release(): card->open_mutex
The patch breaks the chain.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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Upstream-commit-id: df91bc23dcb052ff2da71b3482bf3c5fbf4b8a53
When changing the sample rate, the CMI8788's master clock output becomes
unstable for a short time. The AK4396 needs the master clock to do SPI
writes, so writing to an AK4396 control register directly after a sample
rate change will garble the value. In our case, this leads to the DACs
being misconfigured to I2S sample format, which results in a wrong
output level and horrible distortions on samples louder than -6 dB.
To fix this, we need to wait until the new master clock signal has
become stable before doing SPI writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc upstream
Dell Inspiron 1525 seems to have a buggy BIOS setup and screws up
the recent codec parser, as reported by Oleksandr Natalenko:
http://lkml.org/lkml/2008/9/12/203
This patch adds the working model, dell-3stack, statically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit 3d839e5b87a70effc629c1cdbf77d837ef141919 upstream
Date: Tue, 26 Aug 2008 11:06:26 +0200
Subject: ALSA: oxygen: prevent muting of nonexistent AC97 controls
The Xonar DX does not have CD Capture controls, so we have to check that
a control actually exists before muting it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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Upstream-commit-id: bf9c8c9ddef7ef761ae9747349175adad0ef16ce
If an OSS application calls SNDCTL_DSP_SYNC, then ALSA will call the driver's
_hw_params and _prepare functions again. On the Freescale MPC8610 DMA ASoC
driver, this caused the DMA controller to be unneccessarily re-programmed, and
apparently it doesn't like that. The DMA will then not operate when
instructed. This patch relocates much of the DMA programming to
fsl_dma_open(), which is called only once.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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Upstream-commid-id: be41e941d5f1a48bde7f44d09d56e8d2605f98e1
The Freescale MPC8610 SSI device has the option of using one clock for both
transmit and receive (synchronous mode), or independent clocks (asynchronous).
The SSI driver, however, programs the SSI into synchronous mode and then
tries to program the clock registers independently. The result is that the wrong
sample size is usually generated during recording.
This patch fixes the discrepancy by restricting the sample rate and sample size
of the playback and capture streams. The SSI driver remembers which stream
is opened first. When a second stream is opened, that stream is constrained
to the same sample rate and size as the first stream.
A future version of this driver will lift the sample size restriction.
Supporting independent sample rates is more difficult, because only certain
codecs provide dual independent clocks.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit 82e68f7ffec3800425f2391c8c86277606860442 upstream
snd_seq_oss_synth_make_info() incorrectly reports information
to userspace without first checking for the validity of the
device number, leading to possible information leak (CVE-2008-3272).
Reported-By: Tobias Klein <tk@trapkit.de>
Acked-and-tested-by: Takashi Iwai <tiwai@suse.de>
Cc: stable@kernel.org
Signed-off-by: Willy Tarreau <w@1wt.eu>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit d2cd74b158d7214a556226e3312f9fb1de64d7ae upstream
On Audigy2 Platinum, the Analog/Digital mixer switch is inverted.
https://bugzilla.novell.com/show_bug.cgi?id=396204
The patch adds a simple workaround.
There might be another device requiring a similar fix, too (or fix for
audigy2 generically), but right now I fix only the known broken one.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit 470eaf6be78424fc499a5039e5d5fe58bace2bc3 upstream
Added the missing SSID of Thinkpad Z60m for model=thinkpad with
AD1981HD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit 9ad593f6d326e7a4664e3856520f6c042f82a37f upstream
Many HD-audio controllers seem inaccurate about the IRQ timing of
PCM period updates. This has caused problems on audio quality; e.g.
JACK doesn't work with two periods.
This patch fixes the problem by checking the current DMA position
at IRQ handler and delays the period-update via a workq if it's
inaccurate.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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commit 43785eaeb1cfb8aed3cf8027f298b242f88fdc45 upstream
Don't create mixer volume elements for Headphone and Speaker if they
use the same DAC as normal line-outs on AD1988. Otherwise the amp
value gets screwed up, e.g.
https://bugzilla.novell.com/show_bug.cgi?id=398255
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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Commit 981bcead3f2279a1ec6fb5f2c57aff79ed61a700 upstream.
Stop the S/PDIF DMA engine and output when the device is told to pause.
It will keep on looping the current buffer contents if this isn't done.
Signed-off-by: Pierre Ossman <drzeus@drzeus.cx>
Tested-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Chuck Ebbert <cebbert@redhat.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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snd_assert() in save_mixer() and restore_mixer() in sb_mixer.c is
just wrong. The debug code wasn't tested at all, obviously...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The irq handler may be called before the proper initialization of hardware.
Call snd_aw2_saa7146_setup() before the irq handler registration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Check that model->control_filter is set before trying to call it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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I think that hda_verb array must have "terminator (empty array)".
But alc262_sony_unsol[] does not have it.
And it causes gcc-4.3's buggy behavior
with snd_hda_sequence_write().
Signed-off-by: Akio Idehara <zbe64533@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the Linux kernel is compiled with CONFIG_DEBUG_SHIRQ=y,
the Soundblaster Audigy2 ZS Notebook PCMCIA card causes the
system hang during boot (udev stage) or when the card is hot-plug.
The CONFIG_DEBUG_SHIRQ flag is by default 'y' with all Fedora
kernels since 2.6.23. The problem was reported as
https://bugzilla.redhat.com/show_bug.cgi?id=326411
The issue was hunted down to the snd_emu10k1_create() routine:
/* pseudo-code */
snd_emu10k1_create(...) {
...
request_irq(... IRQF_SHARED ...) {
register the irq handler
#ifdef CONFIG_DEBUG_SHIRQ
call the irq handler: snd_emu10k1_interrupt() {
poll I/O port // <---- !! system hangs
...
}
#endif
}
...
snd_emu10k1_cardbus_init(...) {
initialize I/O ports
}
...
}
The early access to I/O port in the interrupt handler causes
the freeze. Obviously it is necessary to init the I/O ports
before accessing them. This patch moves the registration of
the irq handler after the initialization of the I/O ports.
Signed-off-by: Jaroslav Franek <jarin.franek@post.cz>
Acked-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use quirk table to assign ALC268_TOSHIBA to COMPAL IFL90/JFL-92 laptops.
No analog output on autoprobe.
Signed-off-by: Tony Vroon <tony@linx.net>
Tested-by: Guri <gurashka@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The auto-config mode of Realtek ALC codecs has a bug since 2.6.25
that it cannot resume properly. The problem was the wrong assignment
of init_hook that overrides the whole initialization.
Relevant bug reports:
http://bugzilla.kernel.org/show_bug.cgi?id=10662
https://bugzilla.novell.com/show_bug.cgi?id=385473
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Set the proper model for LG LS75 with CM9880 codec.
See ALSA bug#2105:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2105
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The mic pins are wrongly assigned on AD1884A mobile model.
The mic handling is fixed for the automatic mic selection, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS A9T laptop uses line-out pin as the real front-output while
other devices use it as the surround.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix capture mute widget for STAC9250/9251 codecs. The widget 0x09
has no mute but 0x14 does actually.
Signed-off-by: Mauro Carvalho Chehab <mchehab@infradead.org>
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This solves the problem with mixers wrongly displaying the PWM freq.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added IDs for the Foxconn P35AX-S mainboard to patch_realtek.c, so
that ALC883_6ST_DIG is used by default.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix the missing COEF and EAPD initialization in ALC889 auto-configuration
mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We get quite noisy output on the right channel on VT1708 codec
when 24bit samples are used. Suppress the 24bit support until any
real fix is found.
https://bugzilla.novell.com/show_bug.cgi?id=390473
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added a config table entry for the ASUS P5K-E/WIFI-AP mainboard (ID
1043:8227) to use AD1988_6STACK_DIG
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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Signed-off-by: Al Viro <viro@zeniv.linux.org.uk>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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There is a race from when a device is created with device_create() and
then the drvdata is set with a call to dev_set_drvdata() in which a
sysfs file could be open, yet the drvdata will be NULL, causing all
sorts of bad things to happen.
This patch fixes the problem by using the new function,
device_create_drvdata().
Cc: Kay Sievers <kay.sievers@vrfy.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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Corrected the model assignment for the ASUS P5GD1 w/SPDIF after reports of
surround sound not being possible.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixed the speaker auto-mute with two laptop and docking headphones.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Tony Vroon <tony@linx.net>
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Change HRTIMER_CB_IRQSAFE to HRTIMER_CB_SOFTIRQ,
as suggested by Thomas Gleixner.
That solves the lock dependancy reported in
Bug #10701.
That also allows to call hrtimer_start()
directly, tasklet "stupid hack" removed.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It appears that alsa allows a sound buffer with size not
evenly devided by the period size. This triggers a warning in
snd-pcsp and floods the log. As a quick fix, the warning should
be disabled.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Considering all the feedbacks I got, depending snd-pcsp on
CONFIG_EXPERIMENTAL looks like the only safe way to get out
of all the troubles at one go. :)
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The attached patch adds back the compatibility code, allowing the
driver to work with older alsa-libs.
The removal was premature, it breaks the real-life configs.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added the warning text to the help of snd-pcsp about the possible problem
with this driver so that user can know of the problem in advance.
Also, removed the obsoleted text about ancient pc-speaker patch in
CONFIG_SOUND help.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix this typo and avoid similar errors by using ARRAY_SIZE macro.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is no endpoint called MONOLOUT but MONO_LOUT.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch fixes following bug caught with PREEMPT enabled:
root@b1:~# cat /dev/dsp > /dev/null
BUG: scheduling while atomic: cat/965/0x00000003
Call Trace:
[df165ce0] [c0008e84] show_stack+0x4c/0x1ac (unreliable)
[df165d20] [c001c18c] __schedule_bug+0x64/0x78
[df165d30] [c02b3344] schedule+0x2d8/0x334
[df165d70] [c02b3674] schedule_timeout+0x64/0xe4
[df165db0] [c002c05c] msleep+0x1c/0x34
[df165dc0] [c01f2fe0] fsl_ssi_trigger+0x130/0x144
[df165dd0] [c01ece54] soc_pcm_trigger+0x94/0xb8
[df165df0] [c01da764] snd_pcm_do_start+0x48/0x60
[df165e00] [c01da630] snd_pcm_action_single+0x4c/0xb4
[df165e20] [c01e0f50] snd_pcm_lib_read1+0x2a0/0x2d4
[df165e70] [c01ec274] snd_pcm_oss_read3+0xf0/0x13c
[df165eb0] [c01ec2e4] snd_pcm_oss_read2+0x24/0x4c
[df165ec0] [c01ec4ac] snd_pcm_oss_read+0x1a0/0x1f0
[df165ef0] [c0076478] vfs_read+0xb4/0x108
[df165f10] [c00768cc] sys_read+0x4c/0x90
[df165f40] [c00117a4] ret_from_syscall+0x0/0x38
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When the hardware wavetable synthesizer of an Creative SB Audigy or SB
Live! card (with emu10k chip) receives the MIDI SOFT_PEADAL-press event
(?? 67 127) the appropriate voice is attenuted. Unfortunately when the
pedal is released (event ?? 67 0) the voice does not get it's original
volume again.
Boolean MIDI controls should interpret 0..63 as false and 64..127 as true.
Thanks to Clemens Ladisch for review and correction.
Original patch from "Uwe Kraeger" <uwe_debbug@arcor.de>
Submitted to http://bugs.debian.org/474312
Signed-off-by: maximilian attems <max@stro.at>
Cc: uwe_debbug@arcor.de
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
[ALSA] soc at91 minor bug fixes
[ALSA] soc - at91-pcm - Fix line wrapping
pcspkr: fix dependancies
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drivers/md/raid10.c:889:17: warning: Using plain integer as NULL pointer
drivers/media/video/cx18/cx18-driver.c:616:12: warning: Using plain integer as NULL pointer
sound/oss/kahlua.c:70:12: warning: Using plain integer as NULL pointer
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Cc: Neil Brown <neilb@suse.de>
Cc: Mauro Carvalho Chehab <mchehab@infradead.org>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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Found these two bugs while browsing through the code. The first one is
a cut-n-paste bug, instead of disabling the clock when request_irq()
fails, it enabled it once more. The second one fixes a debug printout,
AT91_SSC_IER is write only, AT91_SSC_IMR is readable (the printed string
actually says imr).
Frank Mandarino was busy so he asked me to send these to this list.
/Patrik
Signed-off-by: Patrik Sevallius <patrik.sevallius@enea.com>
Acked-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There's more checkpatch stuff to fix in the driver, this just fixes the
minimum required for the following patch to be clean.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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fix pcspkr dependancies: make the pcspkr platform
drivers to depend on a platform device, and
not the other way around.
Signed-off-by: Stas Sergeev <stsp@aknet.ru>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Dmitry Torokhov <dtor@mail.ru>
CC: Vojtech Pavlik <vojtech@suse.cz>
CC: Michael Opdenacker <michael-lists@free-electrons.com>
[fixed for 2.6.26-rc1 by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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