| Commit message (Collapse) | Author | Age | Files | Lines |
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Read buffer overflow
ALSA: hda: Correct EAPD for Dell Inspiron 1525
ALSA: hda: warn on spurious response
ALSA: hda: remember last command for each codec
ALSA: hda: read CORBWP inside reg_lock
ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
ALSA: hda: take cmd_mutex in probe_codec()
ALSA: hda: track CIRB/CORB command/response states for each codec
ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
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Check whether index is within bounds before testing the element.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.
The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.
Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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To help disclose hardware bugs.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This converts the last CORBWP access outside of reg_lock.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.
So cmd_mutex would be a safe addition to probe_codec().
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.
Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.
Reference: Novell bnc#526325
https://bugzilla.novell.com/show_bug.cgi?id=526325
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/oss:
sound: mpu401.c: Buffer overflow
sound: aedsp16: Buffer overflow
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mpu_synth_info[m].name is a char[30], and the minimum length of the data
written by sprintf is 31 bytes including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes
including terminating null.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/misc:
ALSA: sound/aoa: Add kmalloc NULL tests
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Check that the result of kzalloc is not NULL before a dereference.
The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
expression *x;
identifier f;
constant char *C;
@@
x = \(kmalloc\|kcalloc\|kzalloc\)(...);
... when != x == NULL
when != x != NULL
when != (x || ...)
(
kfree(x)
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f(...,C,...,x,...)
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*f(...,x,...)
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*x->f
)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/hda:
ALSA: hda - Increase PCM stream name buf in patch_realtek.c
ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
ALSA: hda - Add quirk for Dell Studio 1555
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The name buf with size 16 is too short for some codec names, e.g.
truncated like "ALC861-VD Analo". Now the size is doubled.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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e->sad[] is declared with size ELD_MAX_SAD=16, but the guard
allows range 0-31.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added a quirk entry for Dell Studio 1555.
Reference: Novell bnc#525244
https://bugzilla.novell.com/show_bug.cgi?id=525244
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/usb-audio:
ALSA: usb-audio - Volume control quirk for QuickCam E 3500
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- E3500 report cval->max more than it actually can handel, so if you
set 95% capture level it will be silently muted.
- Betwen cval->min and cval-max(real) is 2940 control units,
but real are only 7 with cval->res = 384.
- Alsa can't handel less than 10 controls, so make it more
and set cval->res = 192.
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/pcm-hwptr:
ALSA: pcm - Fix hwptr buffer-size overlap bug
ALSA: pcm - Fix warnings in debug loggings
ALSA: pcm - Add logging of hwptr updates and interrupt updates
ALSA: pcm - Fix regressions with VMware
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The fix 79452f0a28aa5a40522c487b42a5fc423647ad98 introduced another
bug due to the missing offset for the overlapped hwptr.
When the hwptr goes back to zero, the delta value has to be corrected
with the buffer size. Otherwise this causes looping sounds.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add proper cast.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Added the logging functionality to xrun_debug to record the hwptr
updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(),
corresponding to 16 and 8, respectively.
For example,
# echo 9 > /proc/asound/card0/pcm0p/xrun_debug
will record the position and other parameters at each period interrupt
together with the normal XRUN debugging.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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VMware tends to report PCM positions and period updates at utterly
wrong timing. This screws up the recent PCM core code that tries
to correct the position based on the irq timing.
Now, when a backward irq position is detected, skip the update
instead of rebasing. (This is almost the old behavior before
2.6.30.)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/hda:
ALSA: hda - Fix mute control with some ALC262 models
ALSA: hda - Restore GPIO1 properly at resume with AD1984A
ALSA: hda - Use snprintf() to be safer
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The master mute switch is wrongly implemented as checking the pointer
instead of its value, thus it can be never muted. This patch fixes
the issue.
Reference: Novell bnc#404873
https://bugzilla.novell.com/show_bug.cgi?id=404873
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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The commit 099db17e66294b02814dee01c81d9abbbeece93e introduced a
regression at suspend/resume where the GPIO1 bit isn't properly
restored, thus the speaker output gets muted initially after resume.
The fix is simple, use the cached write for storing GPIO data.
Reference: Novell bnc#522764
https://bugzilla.novell.com/show_bug.cgi?id=522764
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use snprint() for creating the jack name string instead of sprintf()
in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/ctxfi:
ALSA: ctxfi - Fix uninitialized error checks
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Fix a few uninitialized error checks that were introduced recently
mistakenlly during the clean-up:
sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’:
sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’:
sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function
sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’:
sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/caiaq:
ALSA: snd_usb_caiaq: add support for Audio2DJ
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This adds support for Native Instrument's freshly announced Audio2DJ
sound device hardware. Version number bumped to 1.3.19.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/asoc:
ASoC: tlv320aic3x: Enable PLL when not bypassed
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PLL was not being enabled when it was not bypassed. This patch
enables the PLL when it is used. Additionally, it disables the PLL
when it is bypassed.
Without this patch, the audio on TI DM646x EVM and DM355 EVM
does not work properly. The bit clocks and the frame sync signals
from the codec are not correct and hence the playback/record are faster
than usual for most sample rates. The reason for this was that the PLL
was not enabled when it was not bypassed.
Tested on DM6467 EVM, playback tested on DM355 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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* fix/misc:
ALSA: ca0106 - Fix the max capture buffer size
ALSA: OSS sequencer should be initialized after snd_seq_system_client_init
ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
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The capture buffer size with 64kB seems broken with CA0106.
At least, either the update timing or the DMA position is wrong,
and this screws up pulseaudio badly.
This patch restricts the max buffer size less than that to make life
a bit easier.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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When build SND_SEQUENCER in kernel then OSS sequencer(alsa_seq_oss_init)
is initialized before System (snd_seq_system_client_init) which leads to
memory leak :
unreferenced object 0xf6b0e680 (size 256):
comm "swapper", pid 1, jiffies 4294670753
backtrace:
[<c108ac5c>] create_object+0x135/0x204
[<c108adfe>] kmemleak_alloc+0x26/0x4c
[<c1087de2>] kmem_cache_alloc+0x72/0xff
[<c126d2ac>] seq_create_client1+0x22/0x160
[<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
[<c1485a05>] snd_seq_oss_create_client+0x86/0x142
[<c1485920>] alsa_seq_oss_init+0xf6/0x155
[<c1001059>] do_one_initcall+0x4f/0x111
[<c14655be>] kernel_init+0x115/0x166
[<c10032af>] kernel_thread_helper+0x7/0x10
[<ffffffff>] 0xffffffff
unreferenced object 0xf688a580 (size 64):
comm "swapper", pid 1, jiffies 4294670753
backtrace:
[<c108ac5c>] create_object+0x135/0x204
[<c108adfe>] kmemleak_alloc+0x26/0x4c
[<c1087de2>] kmem_cache_alloc+0x72/0xff
[<c126f964>] snd_seq_pool_new+0x1c/0xb8
[<c126d311>] seq_create_client1+0x87/0x160
[<c126e3b6>] snd_seq_create_kernel_client+0x72/0xef
[<c1485a05>] snd_seq_oss_create_client+0x86/0x142
[<c1485920>] alsa_seq_oss_init+0xf6/0x155
[<c1001059>] do_one_initcall+0x4f/0x111
[<c14655be>] kernel_init+0x115/0x166
[<c10032af>] kernel_thread_helper+0x7/0x10
[<ffffffff>] 0xffffffff
unreferenced object 0xf6b0e480 (size 256):
comm "swapper", pid 1, jiffies 4294670754
backtrace:
[<c108ac5c>] create_object+0x135/0x204
[<c108adfe>] kmemleak_alloc+0x26/0x4c
[<c1087de2>] kmem_cache_alloc+0x72/0xff
[<c12725a0>] snd_seq_create_port+0x51/0x21c
[<c126de50>] snd_seq_ioctl_create_port+0x57/0x13c
[<c126d07a>] snd_seq_do_ioctl+0x4a/0x69
[<c126d0de>] snd_seq_kernel_client_ctl+0x33/0x49
[<c1485a74>] snd_seq_oss_create_client+0xf5/0x142
[<c1485920>] alsa_seq_oss_init+0xf6/0x155
[<c1001059>] do_one_initcall+0x4f/0x111
[<c14655be>] kernel_init+0x115/0x166
[<c10032af>] kernel_thread_helper+0x7/0x10
[<ffffffff>] 0xffffffff
The correct order should be :
System (snd_seq_system_client_init) should be initialized before
OSS sequencer(alsa_seq_oss_init) which is equivalent to :
1. insmod sound/core/seq/snd-seq-device.ko
2. insmod sound/core/seq/snd-seq.ko
3. insmod sound/core/seq/snd-seq-midi-event.ko
4. insmod sound/core/seq/oss/snd-seq-oss.ko
Including sound/core/seq/oss/Makefile after other seq modules
fixes the ordering and memory leak.
Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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If spin_lock_irqsave is called twice in a row with the same second
argument, the interrupt state at the point of the second call overwrites
the value saved by the first call. Indeed, the second call does not need
to save the interrupt state, so it is changed to a simple spin_lock.
The semantic match that finds this problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@@
expression lock1,lock2;
expression flags;
@@
*spin_lock_irqsave(lock1,flags)
... when != flags
*spin_lock_irqsave(lock2,flags)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/hda:
ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codecs
ALSA: hda - Add quirk for Gateway T6834c laptop
ALSA: hda_codec: Check for invalid zero connections
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The recent rewrite of the codec parser for STAC9872 caused a regression
for some Sony VAIO models that don't give proper pin default configs
by BIOS. Even using model=vaio doesn't work because the pin definitions
are set after the pin overrides.
This patch fixes the pin definitions in patch_stac9872() to be put
in the right place before the pin overrides. Also the patch adds the
new quirk entry for VAIO F/S to have the correct pin default configs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
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Gateway T6834c laptops need EAPD always on while the default behavior
for the STAC9205 reference board is to turn it off upon every HP plug.
By using the special "eapd" model, which is first introduced for Gateway
T1616 laptops for this same reason, this peculiarity can be properly
handled.
Signed-off-by: Hao Song <baritono.tux@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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To prevent "Too many connections" message and the error path for some HDMI
codecs (which makes onboard audio unusable), check for invalid zero
connections for CONNECT_LIST verb.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/ctxfi:
ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k2
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On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUND
channels were swapped and wrong.
I double checked it with connector colors and creative soundblaster
windows drivers.
So I swapped them to the true order.
Now "speaker-test -c6" and "speaker-test -c8" are working fine.
Signed-off-by: Frank Roth <frashman@freenet.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/usb-audio:
sound: usb-audio: add workaround for Blue Microphones devices
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Blue Microphones USB devices have an alternate setting that sends two
channels of data to the computer. Unfortunately, the descriptors of
that altsetting have a wrong channel setting, which means that any
recorded data from such a device has twice the sample rate from what
would be expected.
This patch adds a workaround to ignore that altsetting. Since these
devices have only one actual channel, no data is lost.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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* fix/misc:
ALSA: riptide - proper handling of pci_register_driver for joystick
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We need to check returning error for pci_register_driver(&joystick_driver)
On failure, we should unregister formerly registered audio drivers
This also fixed the compiler warning :
CC [M] sound/pci/riptide/riptide.o
sound/pci/riptide/riptide.c: In function ‘alsa_card_riptide_init’:
sound/pci/riptide/riptide.c:2200: warning: ignoring return value of ‘__pci_register_driver’, declared with attribute warn_unused_result
Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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