Граф коммитов

92184 Коммитов

Автор SHA1 Сообщение Дата
Mark Brown 815c1be320 [ALSA] pxa2xx-ac97: Support PXA3xx AC97
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:41 +02:00
Mark Brown 7a22323b23 [ALSA] soc - Support PXA3xx AC97
The PXA3xx does not support the use of interrupts during reset and access
to the GPIO status requires similar handling to that for PXA27x.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:41 +02:00
Peter Lienig d80fd0935e [ALSA] ice1712 - Add Terrasoniq TS88 support
Added the support of Terrasonq TS88.

Signed-off-by: Peter Lienig <lienig@rheinmetall-de.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:40 +02:00
Jeff Garzik f000fd8093 [ALSA] Fix synchronize_irq() bugs, redundancies
free_irq() calls synchronize_irq() for you, so there is no need for
drivers to manually do the same thing (again).  Thus, calls where
sync-irq immediately precedes free-irq can be simplified.

However, during this audit several bugs were noticed, where free-irq is
preceded by a "irq >= 0" check... but the sync-irq call is not covered
by the same check.

So, where sync-irq could not be eliminated completely, the missing check
was added.

Signed-off-by: Jeff Garzik <jgarzik@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:40 +02:00
Karsten Wiese 8a87c9cf99 [ALSA] Audiophile 192: Fix ad converter initialization
Correct some arguments in calls to snd_ice1712_gpio_write_bits() from
ap192_set_rate_val().

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:40 +02:00
Karsten Wiese 775c199e6a [ALSA] Don't set gpio mask register in snd_ice1712_gpio_write_bits()
Some calls to snd_ice1712_gpio_write() go wrong, if
snd_ice1712_gpio_write_bits() ran before and changed the gpio mask register.
Read the actual gpio value and combine it with the to be set bits in the cpu
instead.

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:39 +02:00
Karsten Wiese 988f066477 [ALSA] ice1724.c: toggle "chip reset" and "eeprom based setup" sequence
Let "chip reset" become first. Increasement of the "chip reset" related timeout
leads to correctly read eeprom's contents here.

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:39 +02:00
Roel Kluin 0d52cea487 OSS: dmabuf: fix negative DMAbuf_get_buffer_pointer() check
Since unsigned active_offs < 0 is even true when DMAbuf_get_buffer_pointer()
returns negative

Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:39 +02:00
Roel Kluin e34ba21222 [ALSA] SOC: fix tests in cs4270_hw_params()
cs4270_hw_params does several times:

ret = snd_soc_write()
if (ret < 0)
	...

This only works when ret is signed.

Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:39 +02:00
Takashi Iwai 9eb70e68f3 [ALSA] usb-audio - Fix race in reconnection
Fix the race at reconnection of the device.
The disconnected usb_chip[] must be cleared before the next probe
call properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:39 +02:00
Takashi Iwai f18638dcf0 [ALSA] Clean up snd_card_free*()
A little clean up of snd_card_free*().
Removed snd_card_free_prepare() since it's actually almost identical
with snd_card_disconnect().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:39 +02:00
Takashi Iwai 73d38b13ff [ALSA] Fix the race of card instance unregistration
Move the call of device_unregister() for the card instance in
snd_card_disconnect() to avoid the race of sysfs card entry, which
can be typically found on usb-audio reconnection.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:39 +02:00
Risto Suominen 20861fa7b2 [ALSA] snd-powermac: style burgundy.c
Coding style corrections for burgundy.c.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:38 +02:00
Risto Suominen 44deee129c [ALSA] snd-powermac: Burgundy mixers for B&W and iMac
Add mixer controls and correct headphone detection bits for PowerMac
G3 B&W and iMac G3 Tray-loading, both having Burgundy chipset.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:38 +02:00
Risto Suominen 7ae44cfa7a [ALSA] snd-powermac: style awacs.s and awacs.h
Coding style corrections for awacs.c and awacs.h.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:38 +02:00
Risto Suominen a8c2a6bf46 [ALSA] snd-powermac: AWACS and Screamer mixers for PM7500, Beige, and iMac SL
Add mixer controls and correct headphone detection bits for PowerMacs
7300/7500 (AWACS) and G3 Beige (Screamer), and iMac G3 Slot-loading
(Screamer).

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:38 +02:00
Risto Suominen 946cda7d64 [ALSA] snd-powermac: style pmac.c
Coding style corrections for pmac.c.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:38 +02:00
Risto Suominen 9a4f20fcbd [ALSA] snd-powermac: enable headphone detection
Enable port change interrupt while initialising AWACS, Screamer, and
Burgundy chipsets.

Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:38 +02:00
Roel Kluin 369b240d63 [ALSA] sound/drivers/dummy.c: fix negative snd_pcm_format_width() check
bps is unsigned, a negative snd_pcm_format_width() return value is not noticed

Signed-off-by: Roel Kluin <12o3l@tiscali.nl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:38 +02:00
Takashi Iwai 7943a8aba9 [ALSA] hda - Avoid unexpected breakage with ALC889A hack
The last ALC889A hack may break on some devices with certain model presets
since patch_alc*() have different model tables.  So, now it's handled in
the original patch_alc882() but fly to patch_alc883() in model=auto
appropriately.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:37 +02:00
Takashi Iwai cb308f97ae [ALSA] hda - Fix ALC889A codec support
ALC889A is recognized ALC885/ALC882 but it's actually closer to
ALC888/ALC883.

Cc: Kasper Sandberg <lkml@metanurb.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:37 +02:00
Matthew Ranostay 7c2ba97b8a [ALSA] hda: Add 5.1 support for second headphone jack
Several 92hd7xxx and STAC9228 laptops have multiple headphone jacks,
the second headphone jack should be used for the 5.1 surround sound.
Add support for 'Headphone as Line Out' switch, which allows it be used
in 5.1 surround sound.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:37 +02:00
Mark Brown 0a08478c0f [ALSA] soc - wm9712: Remove unneeded AC97_EXTENDED_MID updates
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:37 +02:00
Clemens Ladisch 4972a177fe [ALSA] oxygen: generalize DAC volume TLV handling
Add a pointer for DAC volume TLV data to the model structure so that the
model driver do not need to manually assign it in their control filter.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:37 +02:00
Clemens Ladisch e983532e44 [ALSA] oxygen: mute by default
Initialize the playback volume controls as being muted and having
minimal volume.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:37 +02:00
Clemens Ladisch 193e813814 [ALSA] oxygen: generalize handling of DAC volume limits
Add fields for the DAC volume limits to the module structure so that
model drivers do not need to install their own control info handlers.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:36 +02:00
Clemens Ladisch 0c0e6daf14 [ALSA] hifier: remove empty hifier_mixer_init()
The empty hifier_mixer_init() function is useless; remove it.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:36 +02:00
Takashi Iwai 3adb8abc70 [ALSA] hda - Add support of AD1989A/AD1989B
Added the support of AD1989A and AD1989B codecs.
These codecs can have multiple SPDIF devices, but currently we handle
only one SPDIF.  If any real devices with two SPDIF interfaces (likely
one for SPDIF and one for HDMI), we'll fix this rightly.

Otherwise, these codecs are pretty similar with AD1988.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:36 +02:00
Pavel Machek ad5fada51d [ALSA] sound/core.h: evil #ifdefs
snd_minor_info_oss_* is an function returning int _or_ comment,
depending on config parameters. That is truly evil, fix it.

Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:36 +02:00
Clemens Ladisch a8bb1bad9b [ALSA] virtuoso: fix DX front panel I/O
Fix the GPIO 1 mixer control to enable I/O through the front panel
connector of the Xonar DX.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:36 +02:00
Daniel Mack 6e9fc6bd5d [ALSA] snd_usb_caiaq: make high sample rates work with A8DJ
This patch for snd_usb_caiaq makes sample rates higher dann 48KHz work
with devices which have more than 2 stereo input/output pairs.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:36 +02:00
Daniel Mack 6849d49c48 [ALSA] snd_usb_caiaq: correct input channel order
This patch corrects the input channel order of hardware supported by
snd_usb_caiaq.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:36 +02:00
Daniel Mack 8d048841e8 [ALSA] snd_usb_caiaq: fix potential lockups locking
This patch fixes potential lockups in snd_usb_caiaq by refining the
locking mechanims and by using usb_kill_urb() in favor to
usb_unlink_urb().

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:35 +02:00
Jarkko Nikula f57ab97e76 [ALSA] ASoC: Add support for 19.2 MHz MCLK in TLV320AIC3X
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:35 +02:00
Mark Brown 87b57fe2d3 [ALSA] wm9713: Don't control touch screen power on suspend
Leave the power bit for the touch screen alone when suspending the WM9713
so that the touch screen driver can handle it. This allows the touch
screen to be used as a wakeup source when the system is suspended.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:35 +02:00
Nick Andrew a295e09e89 [ALSA] sound: this amplifier only goes up to 7
sound: kernel log levels are 0-7

Kernel log levels are 0-7, not 0-9.

Signed-off-by: Nick Andrew <nick@nick-andrew.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:35 +02:00
Herton Ronaldo Krzesinski eb5a662166 [ALSA] hda-intel: Add Quanta IL1 ALC267 model
This adds support for Quanta IL1 mini-notebook to alsa, defining a new model
for it. It comes with an ALC267 codec chip. Some notes about this model:

* In headphone automute, I use AC_VERB_SET_PIN_WIDGET_CONTROL instead of common
  amp mute, to avoid conflict with mixer switch (mixer and automute use the
  same nid).
* The only connected capture sources in the hardware are the internal mic and
  external mic jack. So instead of using an input source selector like on other
  ALC268 models, the mic automute automatically switch between captures.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:35 +02:00
Kay Sievers 8b45a20993 [ALSA] sound: fix platform driver hotplug/coldplug
Since 43cc71eed1, the platform modalias is
prefixed with "platform:".  Add MODULE_ALIAS() to the hotpluggable sound
platform drivers, to re-enable auto loading.

[dbrownell@users.sourceforge.net: more drivers, registration fixes]

Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:35 +02:00
Matthew Ranostay 0fc9dec46f [ALSA] hda: EAPD power management
Power management support for EAPD enabled laptops, when headphones
are sensed it pulls the EAPD GPIO line low to power it down.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:35 +02:00
Matthew Ranostay 780c8be4ab [ALSA] hda: Correct SPDIF out default config
Several laptops have have the SPDIF out defined as 'Digital other out'
when it should be 'SPDIF out' in the default config.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:34 +02:00
Tony Vroon 06a9c30cdd [ALSA] hda - Fujitsu Lifebook PC speaker signal
The legacy PC speaker signal was not routed to outputs. The codec is not
prevented from powering down in this patch, although I suppose one could
argue that perhaps it should be. Let me know if anyone feels strongly one
way or the other.

Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:34 +02:00
Jiang zhe 5b030389e4 [ALSA] hda - PCI quirk for laptop LG which use CMI9880
Please refer to [0003874] on the alsa mantis.
This patch added the pci quirk.

Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:34 +02:00
Jiang zhe 64654c2f9e [ALSA] hda - Should use HDA_OUTPUT instead of HDA_INPUT to mute pin 15 of ALC880
To mute the output of Pin widget 15 in ALC880, we should use the
HDA_OUTPUT. However, current code looks like :
snd_hda_codec_amp_stereo(codec, 0x15, HDA_INPUT, 0, HDA_AMP_MUTE, bits);
It may be a misspelling.

Signed-off-by: Jiang zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:34 +02:00
Pavel Machek 07f51a7274 [ALSA] sound/usb/usbaudio.c: coding style
Putting space between ! and variable is a strange coding style, fix
that, also make it fit into 80 columns where that is easy.

Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:34 +02:00
Pavel Machek 2a56f51bcc [ALSA] usb audio: make quirk handling more readable, and fix commented-out code
usb audio contains useful  debugging code, protected by #if
0. Unfortunately, it will not compile because variable names changed;
fix it.

Dallas workaround is formatted in a way where it is not quite obvious
what is normal code and what is quirk. Reformat it to make it obvious.

Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:34 +02:00
Pavel Machek b9d43bcd06 [ALSA] usb audio: Fix another Dallas quirk
Dallas USB speakers are buggy in more than one way. One of configs
they offer does not work at all.

Signed-off-by: Pavel Machek <pavel@suse.cz>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:33 +02:00
Frederik Deweerdt eaa9985b4e [ALSA] hda-codec - Fix unbalanced mutex
On Wed, Apr 02, 2008 at 08:19:29AM -0400, Miles Lane wrote:
> [   48.765906] [ BUG: bad unlock balance detected! ]
> [   48.765912] -------------------------------------
> [   48.765918] pulseaudio/4277 is trying to release lock
> (&codec->spdif_mutex) at:
> [   48.765930] [<c03031b7>] mutex_unlock+0x8/0xa
> [   48.765945] but there are no more locks to release!
> [   48.765950]
> [   48.765952] other info that might help us debug this:
> [   48.765959] 2 locks held by pulseaudio/4277:
> [   48.765965]  #0:  (&pcm->open_mutex){--..}, at: [<f89f134b>]
> snd_pcm_open+0xc1/0x1ba [snd_pcm]
> [   48.766003]  #1:  (&chip->open_mutex){--..}, at: [<f8b4f13d>]
> azx_pcm_open+0x36/0x184 [snd_hda_intel]
> [   48.766057]
> [   48.766059] stack backtrace:
> [   48.766066] Pid: 4277, comm: pulseaudio Not tainted 2.6.25-rc8-mm1 #12
> [   48.766086]  [<c013afc6>] print_unlock_inbalance_bug+0xce/0xd8
> [   48.766107]  [<c0109e1c>] ? save_stack_trace+0x1d/0x3b
> [   48.766130]  [<c012f54e>] ? __kernel_text_address+0x1b/0x27
> [   48.766146]  [<c0104533>] ? dump_trace+0xcd/0xd9
> [   48.766160]  [<c0109d9e>] ? save_stack_address+0x0/0x2c
> [   48.766176]  [<c013b80a>] ? find_usage_backwards+0xa4/0xc3
> [   48.766193]  [<c013cfb5>] lock_release_non_nested+0x84/0x120
> [   48.766209]  [<c03031b7>] ? mutex_unlock+0x8/0xa
> [   48.766222]  [<c013d1bb>] lock_release+0x16a/0x199
> [   48.766238]  [<c0303137>] __mutex_unlock_slowpath+0xa9/0x121
> [   48.766252]  [<c03031b7>] mutex_unlock+0x8/0xa
> [   48.766263]  [<f8b4ffd8>] snd_hda_multi_out_analog_open+0xd3/0xef
> [snd_hda_intel]

The following patch should fix it.

Cc: "Miles Lane" <miles.lane@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:33 +02:00
Andrew Morton 66c9aa6043 [ALSA] es1968 - fix coding style in the last patch
WARNING: braces {} are not necessary for single statement blocks
#40: FILE: sound/pci/es1968.c:1831:
+       if (diff > 1) {
+               __maestro_write(chip, IDR0_DATA_PORT, cp1);
+       }

total: 0 errors, 1 warnings, 35 lines checked

./patches/es1968-fix-jitter-on-some-maestro-cards.patch has style problems, please review.  If any of these errors
are false positives report them to the maintainer, see
CHECKPATCH in MAINTAINERS.

Please run checkpatch prior to sending patches

Cc: Andreas Mueller <andreas@stapelspeicher.org>
Tested-by: Rene Herman <rene.herman@keyaccess.nl>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:33 +02:00
Andreas Mueller f24bfa53da [ALSA] es1968: fix jitter on some maestro cards
This patch suppresses jitter on several Maestro cards in stereo mode (ALSA of
course).

The patch is also incorporated in the *BSD drivers where I "ported" it from.

Without this patch most of the stereo audio gets out of sync and really
distorted (oss-emulation with mplayer at 48000khz worked somehow).

Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:33 +02:00
Denys Vlasenko 62cef8212f [ALSA] sound/pci/rme9652/hdspm.c: stop inlining largish static functions
sound/pci/rme9652/hdspm.c has unusually large number of static inline
functions - 22.

I looked through them and some of them seem to be too big to warrant inlining.

This patch removes "inline" from these static functions (regardless of number
of callsites - gcc nowadays auto-inlines statics with one callsite).

Size difference on 32bit x86:
   text    data     bss     dec     hex filename
  20437    2160     516   23113    5a49 linux-2.6-ALLYES/sound/pci/rme9652/hdspm.o
  18036    2160     516   20712    50e8 linux-2.6.inline-ALLYES/sound/pci/rme9652/hdspm.o

[coding fix by Takashi Iwai <tiwai@suse.de>]

Signed-off-by: Denys Vlasenko <vda.linux@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-04-24 12:00:33 +02:00