| Commit message (Collapse) | Author | Age |
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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: HDA: Realtek: Avoid unnecessary volume control index on Surround/Side
ASoC: Support !REGULATOR build for sgtl5000
ALSA: hda - VIA: Fix VT1708 can't build up Headphone control issue
ALSA: hda - VIA: Correct stream names for VT1818S
ALSA: hda - VIA: Fix codec type for VT1708BCE at the right timing
ALSA: hda - VIA: Fix invalid A-A path volume adjust issue
ALSA: hda - VIA: Add missing support for VT1718S in A-A path
ALSA: hda - VIA: Fix independent headphone no sound issue
ALSA: hda - VIA: Fix stereo mixer recording no sound issue
ALSA: hda - Set EAPD for Realtek ALC665
ALSA: usb - Remove trailing spaces from USB card name strings
sound: read i_size with i_size_read()
ASoC: Remove bogus check for register validity in debugfs write
ASoC: mini2440: Fix uda134x codec problem.
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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>
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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>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.39
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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
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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: (308 commits)
ALSA: sound/pci/asihpi: check adapter index in hpi_ioctl
ALSA: aloop - Fix possible IRQ lock inversion
ALSA: sound/core: merge list_del()/list_add_tail() to list_move_tail()
ALSA: ctxfi - use list_move() instead of list_del()/list_add() combination
ALSA: firewire - msleep needs delay.h
ALSA: firewire-lib, firewire-speakers: handle packet queueing errors
ALSA: firewire-lib: allocate DMA buffer separately
ALSA: firewire-lib: use no-info SYT for packets without SYT sample
ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver
ALSA: hda - Remove an unused variable in patch_realtek.c
ALSA: hda - pin-adc-mux-dmic auto-configuration of 92HD8X codecs
ALSA: hda - fix digital mic selection in mixer on 92HD8X codecs
ALSA: hda - Move default input-src selection to init part
ALSA: hda - Initialize special cases for input src in init phase
ALSA: ctxfi - Clear input settings before initialization
ALSA: ctxfi - Fix SPDIF status retrieval
ALSA: ctxfi - Fix incorrect SPDIF status bit mask
ALSA: ctxfi - Fix microphone boost codes/comments
ALSA: atiixp - Fix wrong time-out checks during ac-link reset
ALSA: intel8x0m: append 'm' to "r_intel8x0"
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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>
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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>
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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>
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Signed-off-by: Kirill A. Shutemov <kirill@shutemov.name>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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- 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>
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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>
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Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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Depending on the model and the presence of a TCO module, the number of
midi ports varies. Some have 1 port (MADIface), some have 2 (default),
with TCO, there are 3.
Don't hardcode the number of midi ports to initialize.
This patch also fixes a boot lockup on MADIface.
[Coding-style fixes by tiwai]
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Don't continue if we cannot detect the real card type, otherwise, all
subsequent functions, especially strcpy(), would fail, leaving the whole
driver in an unusable state.
Without such a protection, dmesg would look like this:
Pid: 1525, comm: modprobe Not tainted 2.6.38-rc6 #1 MSI MS-7250/MS-7250
EIP: 0060:[<c114e700>] EFLAGS: 00010296 CPU: 1
EIP is at strcpy+0x10/0x30
EAX: f4d33e58 EBX: f5990800 ECX: f4d33e58 EDX: 00000000
ESI: 00000000 EDI: f4d33e58 EBP: f5990930 ESP: f3dd3e0c
DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068
f53d4204 00000000 f90bfc03 00000001 00000001 f3dd3e64 c10fedff f3dd3e7c
f3d854e4 f53d4000 c10307b8 00000001 c10307b8 f5990860 c10307b8 00000001
c10332e5 f59908bc c12f4339 f59908bc f5990860 c11ef207 f4d33e00 f53d4000
[<f90bfc03>] ? snd_hdspm_probe+0x7e7/0x1166 [snd_hdspm]
[<c10fedff>] ? __sysfs_add_one+0x1f/0xf0
[<c10307b8>] ? get_parent_ip+0x8/0x20
[<c10307b8>] ? get_parent_ip+0x8/0x20
[<c10307b8>] ? get_parent_ip+0x8/0x20
[<c10332e5>] ? add_preempt_count+0xa5/0xd0
[<c12f4339>] ? _raw_spin_lock_irqsave+0x19/0x40
[<c11ef207>] ? pm_runtime_enable+0x17/0x80
[<c1160172>] ? local_pci_probe+0x42/0xb0
[<c1161091>] ? pci_device_probe+0x61/0x80
[<c11e9587>] ? driver_probe_device+0x77/0x180
[<c11600f0>] ? pci_match_device+0xa0/0xc0
[<c11e9709>] ? __driver_attach+0x79/0x80
[<c11e9690>] ? __driver_attach+0x0/0x80
[<c11e8e32>] ? bus_for_each_dev+0x52/0x80
[<c11e9416>] ? driver_attach+0x16/0x20
[<c11e9690>] ? __driver_attach+0x0/0x80
[<c11e8771>] ? bus_add_driver+0xa1/0x220
[<c1160f20>] ? pci_device_remove+0x0/0xf0
[<c11e99a3>] ? driver_register+0x63/0x120
[<c11612bd>] ? __pci_register_driver+0x3d/0xb0
[<c1001132>] ? do_one_initcall+0x32/0x160
[<f90d6000>] ? alsa_card_hdspm_init+0x0/0x14 [snd_hdspm]
[<c10676a9>] ? sys_init_module+0x99/0x1e0
[<c10ade3d>] ? sys_close+0x6d/0xc0
[<c1002c90>] ? sysenter_do_call+0x12/0x26
---[ end trace 239f0661c558378b ]---
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In contrast to the PCIe version (RME AES), the PCI version (RME AES32)
has a different firmware revision.
This patch adds the missing PCI revision.
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Allow input from microphone on remaining PowerBooks and iBooks with Tumbler.
Tested only with PowerBook G4 Gigabit Ethernet (PowerBook3,3).
[Fixed to use of_machine_is_compatible by tiwai]
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Patch below removes one to many "n's" in a word..
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The patch below removes an extra "l" in the word.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The patch below removes an extra "l" in the word.
Signed-off-by: Justin P. Mattock <justinmattock@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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