openwrt/staging/blogic.git
16 years ago[ALSA] hda-intel - Add workarounds for STAC codecs
Takashi Iwai [Wed, 16 Jan 2008 15:09:47 +0000 (16:09 +0100)]
[ALSA] hda-intel - Add workarounds for STAC codecs

Some machines with STAC codecs seem to have problems (e.g. no audible
playback) when the delay in codec-read routine is too short.
I still don't figure out which command sequence causes this problem
(due to lack of test hardware), but it's known that increasing the
delay fixes.  So, added a stupid workaround here temporarily...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hdsp: make Multiface II work again
Andreas Degert [Wed, 16 Jan 2008 14:59:48 +0000 (15:59 +0100)]
[ALSA] hdsp: make Multiface II work again

This device has io_type == 1 (Multiface) and firmware_rev > 0xa
(fixes regression from changeset 5326)

Signed-off-by: Andreas Degert <ad@papyrus-gmbh.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] snd-powermac: handle dead DMA transfers
T. H. Huth [Wed, 16 Jan 2008 14:57:08 +0000 (15:57 +0100)]
[ALSA] snd-powermac: handle dead DMA transfers

This patch provides the snd-powermac sound driver with the ability to handle
dead DMA transfers. If a dead DMA transfer is detected, the driver now sets
up a new DMA transfer to continue with the sound output at the point where the
old transfer died.
This dead DMA transfer handling has become necessary with recent kernels on
certain G4 PowerMacs. Please refer to the following URLs for more information:
 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3126
 https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/87652
 http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=436723
The patch is based on the dead DMA transfer handling code from the old dmasound
driver which can be found in the file sound/oss/dmasound/dmasound_awacs.c in
the Linux source code.

Signed-off-by: T. H. Huth <th.huth@googlemail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound: fix caiaq section mismatches
Randy Dunlap [Wed, 16 Jan 2008 13:56:04 +0000 (14:56 +0100)]
[ALSA] sound: fix caiaq section mismatches

Fix section mismatch in caiaq: these __devinit functions can be
called at any time so they should not be __devinit.
WARNING: vmlinux.o(.text+0x10a8dae): Section mismatch: reference to .init.text:snd_usb_caiaq_audio_init (between 'setup_card' and 'create_card')
WARNING: vmlinux.o(.text+0x10a8dd6): Section mismatch: reference to .init.text:snd_usb_caiaq_midi_init (between 'setup_card' and 'create_card')

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound: fix rme9652 section mismatch
Randy Dunlap [Wed, 16 Jan 2008 13:55:42 +0000 (14:55 +0100)]
[ALSA] sound: fix rme9652 section mismatch

Fix section mismatch in hdsp:  snd_hdsp_proc_init() can be called from
an ioctl at any time.
WARNING: vmlinux.o(.text+0x1089bc2): Section mismatch: reference to .init.text: (between 'snd_hdsp_create_alsa_devices' and 'snd_hdsp_free')

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound: fix atiixp section mismatch
Randy Dunlap [Wed, 16 Jan 2008 13:55:07 +0000 (14:55 +0100)]
[ALSA] sound: fix atiixp section mismatch

Fix section mismatch in atiixp by making some functions __devinit.
WARNING: vmlinux.o(.text+0xfd9304): Section mismatch: reference to .init.data:atiixp_quirks (between 'ac97_probing_bugs' and 'snd_atiixp_codec_detect')

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound: fix ad1889 section mismatch
Randy Dunlap [Wed, 16 Jan 2008 13:54:46 +0000 (14:54 +0100)]
[ALSA] sound: fix ad1889 section mismatch

Fix section mismatch in ad1889 by renaming the pci_driver variable to a
whitelisted variable name.
WARNING: vmlinux.o(.data+0x2e5ff0): Section mismatch: reference to .init.text:snd_ad1889_probe (between 'ad1889_pci' and 'index')

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound: fix mts64 section mismatches
Randy Dunlap [Wed, 16 Jan 2008 13:54:21 +0000 (14:54 +0100)]
[ALSA] sound: fix mts64 section mismatches

Fix section mismatches in mts64 by making a static variable __devinitdata.
WARNING: vmlinux.o(.data+0x2e33f0): Section mismatch: reference to .init.data:mts64_ctl_smpte_switch (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e33f8): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_hours (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3400): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_minutes (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3408): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_seconds (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3410): Section mismatch: reference to .init.data:mts64_ctl_smpte_time_frames (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')
WARNING: vmlinux.o(.data+0x2e3418): Section mismatch: reference to .init.data:mts64_ctl_smpte_fps (between 'control.19929' and 'snd_mts64_rawmidi_output_ops')

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: make PCM limits configurable
Clemens Ladisch [Wed, 16 Jan 2008 07:32:53 +0000 (08:32 +0100)]
[ALSA] oxygen: make PCM limits configurable

Add a callback to the model structure to allow modification of the
hardware PCM limits.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: add control filter to model struct
Clemens Ladisch [Wed, 16 Jan 2008 07:32:08 +0000 (08:32 +0100)]
[ALSA] oxygen: add control filter to model struct

Allow the models to modify mixer controls before they are added to the
card.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: make all DMA channels configurable
Clemens Ladisch [Wed, 16 Jan 2008 07:30:38 +0000 (08:30 +0100)]
[ALSA] oxygen: make all DMA channels configurable

Allow the card models to specify whether each of the hardware DMA
channels is used.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: make SPI configuration configurable
Clemens Ladisch [Wed, 16 Jan 2008 07:28:54 +0000 (08:28 +0100)]
[ALSA] oxygen: make SPI configuration configurable

Add a field to the model structure so that it is possible to have a card
where the SPI outputs 4 and 5 are used for an EEPROM.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: make AC97 codec optional
Clemens Ladisch [Wed, 16 Jan 2008 07:28:17 +0000 (08:28 +0100)]
[ALSA] oxygen: make AC97 codec optional

Only initialize and create mixer controls for the first AC97 codec when
one has actually been detected.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Enable VIA SPDIF input pin
Takashi Iwai [Tue, 15 Jan 2008 11:39:38 +0000 (12:39 +0100)]
[ALSA] hda-codec - Enable VIA SPDIF input pin

Enable the SPDIF input-pin on VIA codecs when SPDIF-input is enabled
by BIOS.  Also, including a bit code clean up.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Add the support of Dell OEM laptops with ALC268
Takashi Iwai [Tue, 15 Jan 2008 11:37:42 +0000 (12:37 +0100)]
[ALSA] hda-codec - Add the support of Dell OEM laptops with ALC268

Added the support of Dell OEM laptops (Vostro 1200) with ALC268 codec.
The new model=dell is provided.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Disable PCBEEP mixer element in test model
Takashi Iwai [Tue, 15 Jan 2008 10:41:41 +0000 (11:41 +0100)]
[ALSA] hda-codec - Disable PCBEEP mixer element in test model

It turned out that the PCBEEP element (0x1d) is disabled on some hardwares
although it's defined in the datasheet.  Because of the error at info of
this element, the mixer gets totally unusable.
Since the PCBEEP isn't that important feature, it's safer to disable this.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-code - Clean up STAC GPIO enablement code
Takashi Iwai [Tue, 15 Jan 2008 10:39:08 +0000 (11:39 +0100)]
[ALSA] hda-code - Clean up STAC GPIO enablement code

There are two similar GPIO-enablement codes in patch_sigmatel.c.
Let's clean up.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] HDA: Enable chipset gcap usage
Tobin Davis [Tue, 15 Jan 2008 10:23:55 +0000 (11:23 +0100)]
[ALSA] HDA: Enable chipset gcap usage

This patch removes hardcoded values for the number of streams supported
by the southbridge in most chipsets, and reads these values from the
chipset directly.  Most systems are hardwired for 4 streams in each
direction, but newer chipsets change that capability.

Signed-off-by: Tobin Davis <tdavis@dsl-only.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: rename PCM to Master
Clemens Ladisch [Tue, 15 Jan 2008 07:39:06 +0000 (08:39 +0100)]
[ALSA] oxygen: rename PCM to Master

Rename the 'PCM Playback Volume'/'Switch' mixer controls to 'Master'.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] cs4231: remove one busy wait
Krzysztof Helt [Mon, 14 Jan 2008 11:07:53 +0000 (12:07 +0100)]
[ALSA] cs4231: remove one busy wait

This busy_wait is not needed after latest changes
to the cs4231-lib

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: make line-in switch exclusive
Clemens Ladisch [Mon, 14 Jan 2008 07:57:05 +0000 (08:57 +0100)]
[ALSA] oxygen: make line-in switch exclusive

The line input cannot be mixed with the other inputs, so we have to mute
the other input switches when it is selected.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: use an array of snd_kcontrol pointers
Clemens Ladisch [Mon, 14 Jan 2008 07:56:01 +0000 (08:56 +0100)]
[ALSA] oxygen: use an array of snd_kcontrol pointers

Use an array for the pointers to known controls so that it is easier to
add more.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: fix channel routing
Clemens Ladisch [Mon, 14 Jan 2008 07:55:03 +0000 (08:55 +0100)]
[ALSA] oxygen: fix channel routing

Do not exchange the surround and back jacks except when in 7.1 mode
where the surround jack is not rear but side.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] fix compilation warning in GCC
Miguel Boton [Sun, 13 Jan 2008 11:03:53 +0000 (12:03 +0100)]
[ALSA] fix compilation warning in GCC

'snd_shutdown_f_ops' is not a pointer so its address will never be NULL.
GCC will complain because 'fops_get' will do an unnecessary check because
'&snd_shutdown_f_ops' is always true.

Signed-off-by: Miguel Boton <mboton@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Remove obsolete FIXME's
Takashi Iwai [Sun, 13 Jan 2008 11:03:05 +0000 (12:03 +0100)]
[ALSA] hda-codec - Remove obsolete FIXME's

Removed 'FIXME' comments that have been already fixed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: Fix 5.1 sound in Dell 6stack ALC888 HDA
Claudio Matsuoka [Sun, 13 Jan 2008 10:58:27 +0000 (11:58 +0100)]
[ALSA] hda: Fix 5.1 sound in Dell 6stack ALC888 HDA

This patch fixes 5.1 surround output and headphone detection in the
Dell Inspiron 530 and possibly other Dell systems using the ALC888
codec (mode 6stack-dell).

Signed-off-by: Claudio Matsuoka <cmatsuoka@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda_intel: Fix multiple device support by incrementing device count
Andrew Paprocki [Sun, 13 Jan 2008 10:57:17 +0000 (11:57 +0100)]
[ALSA] hda_intel: Fix multiple device support by incrementing device count

Fixes multiple device support by incrementing the static device counter
at the end of the azx_probe() call. Without this, subsequent probes would
always use the index specified for the first card.

Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Add ASoC drivers for the Freescale MPC8610 SoC
Timur Tabi [Fri, 11 Jan 2008 17:15:26 +0000 (18:15 +0100)]
[ALSA] Add ASoC drivers for the Freescale MPC8610 SoC

Add the ASoC drivers for the Freescale MPC8610 SoC and the MPC8610 HPCD
reference board.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Don't build boost controls for digital mics
Takashi Iwai [Fri, 11 Jan 2008 16:38:35 +0000 (17:38 +0100)]
[ALSA] hda-codec - Don't build boost controls for digital mics

The ALC auto-probe creates mic boost controls automatically for the
probed pins, but it assumes that they are analog mics.  The digital
mics have no boost controls and must be skipped.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - print control name in error messages
Takashi Iwai [Fri, 11 Jan 2008 15:12:23 +0000 (16:12 +0100)]
[ALSA] hda-codec - print control name in error messages

Print the name of the defect controls in error messages in amp info
callback.  This will make debugging easier.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: Add new STAC9205 PCI_QUIRK
Matthew Ranostay [Fri, 11 Jan 2008 10:39:06 +0000 (11:39 +0100)]
[ALSA] hda: Add new STAC9205 PCI_QUIRK

Added a new STAC 9205 quirk for Vostro 1500.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] PCM interface - rename SNDRV_PCM_TSTAMP_MMAP to SNDRV_PCM_TSTAMP_ENABLE
Jaroslav Kysela [Fri, 11 Jan 2008 07:45:08 +0000 (08:45 +0100)]
[ALSA] PCM interface - rename SNDRV_PCM_TSTAMP_MMAP to SNDRV_PCM_TSTAMP_ENABLE

Change semantics for SNDRV_PCM_TSTAMP_MMAP. Doing timestamping only in
the interrupt handler might cause that hw_ptr is not related to actual
timestamp. With this change, grab timestamp at every hw_ptr update to
have always valid timestamp + ring buffer position pair.
With this change, SNDRV_PCM_TSTAMP_MMAP was renamed to
SNDRV_PCM_TSTAMP_ENABLE. It's no regression (I think).

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: 92HD7XXX power management support
Matthew Ranostay [Thu, 10 Jan 2008 15:55:06 +0000 (16:55 +0100)]
[ALSA] hda: 92HD7XXX power management support

Added support for advanced power management support for output ports on
92HD7xxx family of codecs. Inactive output ports are powered down when
the pin sense  doesn't detect a connection, and powered back up when a
connection is sensed.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Add virtual master controls
Takashi Iwai [Thu, 10 Jan 2008 15:53:55 +0000 (16:53 +0100)]
[ALSA] hda-codec - Add virtual master controls

Add master controls using vmaster to codecs that have no real hardware
master volume registers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Add virtual master control helpers
Takashi Iwai [Thu, 10 Jan 2008 15:52:42 +0000 (16:52 +0100)]
[ALSA] Add virtual master control helpers

Added helper functions to implement virtual master volume controls.
The virtual master control is a control element that has multiple
slave controls.  The value of master element is equally added to
slave elements.
The functions are written for general purpose, but it's put in the
HD-audio directory as now, since HD-audio driver is the only user.
It should be moved to the common place once after other drivers use
vmaster.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Preliminary ac97 drivers for Toshiba e800 PDAs
Ian Molton [Thu, 10 Jan 2008 13:50:34 +0000 (14:50 +0100)]
[ALSA] soc - Preliminary ac97 drivers for Toshiba e800 PDAs

Currently only the AUX channel is used (touchscreen)

Signed-off-by: Ian Molton <spyro@f2s.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] S3C2412: suspend and resume support
Ben Dooks [Thu, 10 Jan 2008 13:48:37 +0000 (14:48 +0100)]
[ALSA] S3C2412: suspend and resume support

Support for suspend/resume for the S3C2412 ASoC IIS
core driver.

Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] ASoC: S3C2412 IIS driver
Ben Dooks [Thu, 10 Jan 2008 13:47:21 +0000 (14:47 +0100)]
[ALSA] ASoC: S3C2412 IIS driver

S3C2412 SoC IIS support for ALSA/ASoC

Signed-off-by: Ben Dooks <ben-linux@fluff.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Support suspend and resume of the I2S interface on s3c24xx
Graeme Gregory [Thu, 10 Jan 2008 13:44:58 +0000 (14:44 +0100)]
[ALSA] soc - Support suspend and resume of the I2S interface on s3c24xx

Signed-off-by: Graeme Gregory <graeme@openmoko.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Reinitialise DMA on every resume
Graeme Gregory [Thu, 10 Jan 2008 13:44:24 +0000 (14:44 +0100)]
[ALSA] soc - Reinitialise DMA on every resume

This one changes the DMA initialisation as it turns out the DMA driver
in s3c24xx doesnt store registers between suspend/resume so you have
to re-initialise the channels on every resume.

Signed-off-by: Graeme Gregory <graeme@openmoko.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Initial WM8753 TLV support for capture mixer
Liam Girdwood [Thu, 10 Jan 2008 13:43:48 +0000 (14:43 +0100)]
[ALSA] soc - Initial WM8753 TLV support for capture mixer

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Bump ASoC core version number
Mark Brown [Thu, 10 Jan 2008 13:53:48 +0000 (14:53 +0100)]
[ALSA] Bump ASoC core version number

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Don't lock the codec list in snd_soc_dapm_new_widgets()
Mark Brown [Thu, 10 Jan 2008 13:41:46 +0000 (14:41 +0100)]
[ALSA] soc - Don't lock the codec list in snd_soc_dapm_new_widgets()

snd_soc_dapm_new_widgets() takes the codec lock when adding new widgets,
causing lockdep warnings when applications later call down through ALSA
to adjust controls.  Since widgets are only added during probe this lock
should be unneeded so don't take it.
Thanks to Dmitry Baryshkov <dbaryshkov@gmail.com> for reporting this issue.
Cc: Dmitry Baryshkov <dbaryshkov@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Add support for passing kcontrols with events
Laim Girdwood [Thu, 10 Jan 2008 13:41:02 +0000 (14:41 +0100)]
[ALSA] soc - Add support for passing kcontrols with events

Signed-off-by: Laim Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Clean up tabs
Liam Girdwood [Thu, 10 Jan 2008 13:40:16 +0000 (14:40 +0100)]
[ALSA] soc - Clean up tabs

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Fix power switching support for DAPM_SWITCH widgets
Milan plzik [Thu, 10 Jan 2008 13:39:46 +0000 (14:39 +0100)]
[ALSA] soc - Fix power switching support for DAPM_SWITCH widgets

Signed-off-by: Milan plzik <milan.plzik@gmail.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Ensure PCMs are suspended
Liam Girdwood [Thu, 10 Jan 2008 13:39:01 +0000 (14:39 +0100)]
[ALSA] soc - Ensure PCMs are suspended

This fixes a bug whereby PCMs were not being suspended when the rest of the
audio subsystem was suspended.

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Add D1 power event to power down event sequence
Liam Girdwood [Thu, 10 Jan 2008 13:38:24 +0000 (14:38 +0100)]
[ALSA] soc - Add D1 power event to power down event sequence

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] ASoC TLV support
Philipp Zabel [Thu, 10 Jan 2008 13:37:42 +0000 (14:37 +0100)]
[ALSA] ASoC TLV support

Add TLV support to ASoC.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] soc - Add device level DAPM event
Liam Girdwood [Thu, 10 Jan 2008 13:36:20 +0000 (14:36 +0100)]
[ALSA] soc - Add device level DAPM event

Added a device level dapm event so that both the machine and codec are informed
when dapm events occur.

Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Fix inverted Phone volume WM9712 mixer control
Joe Sauer [Thu, 10 Jan 2008 13:34:56 +0000 (14:34 +0100)]
[ALSA] Fix inverted Phone volume WM9712 mixer control

Signed-off-by: Joe Sauer <jsauer@vernier.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Update MAINTAINERS for ALSA SoC
Mark Brown [Thu, 10 Jan 2008 13:33:07 +0000 (14:33 +0100)]
[ALSA] Update MAINTAINERS for ALSA SoC

Add myself as a point of contact for the ALSA SoC subsystem and add a
reference to the development GIT tree.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: STAC9228 VT fixes
Matthew Ranostay [Thu, 10 Jan 2008 12:06:26 +0000 (13:06 +0100)]
[ALSA] hda: STAC9228 VT fixes

Moved 2 systems PCI_QUIRK values to STAC_DELL_BIOS. Also the second
front HP jack is incorrect defined in the BIOS VT's for some laptops,
this patch corrects this.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Device ID for Macbook sound card
Jiang zhe [Thu, 10 Jan 2008 12:05:47 +0000 (13:05 +0100)]
[ALSA] hda-codec - Device ID for Macbook sound card

Please refer to the [0003680] on ALSA bugtracking system.
The user found that 'model=mbp3' works and provided the ID.
From: Jiang zhe <zhe.jiang@intel.com>

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Update realtek codec support
Kailang Yang [Thu, 10 Jan 2008 12:03:59 +0000 (13:03 +0100)]
[ALSA] hda-codec - Update realtek codec support

1. Support HP rp5700
2. Fixed alc_subsystem_id function (Bug fixed and support Desktop)
3. Support ASUS EP20

Signed-off-by: Kailang Yang <kailang@realtek.com.tw>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] rawmidi: let sparse know what is going on _for real_
Marcin Ślusarz [Wed, 9 Jan 2008 16:56:07 +0000 (17:56 +0100)]
[ALSA] rawmidi: let sparse know what is going on _for real_

snd_rawmidi_kernel_read1/write1 weren't annotated but used
copy_to_user/copy_from_user when one of parameters (kernel) was equal to 0
remove it and add properly annotated parameter

Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: 92HD71BXX Mono Mute Support
Matthew Ranostay [Wed, 9 Jan 2008 11:30:20 +0000 (12:30 +0100)]
[ALSA] hda: 92HD71BXX Mono Mute Support

Added a mono output mute mixer for the 92hd71bxx family of codecs, this
also removes the need for the mono out node to explicitly unmuted in the
core init.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Remove sound/driver.h
Takashi Iwai [Tue, 8 Jan 2008 17:13:27 +0000 (18:13 +0100)]
[ALSA] Remove sound/driver.h

This header file exists only for some hacks to adapt alsa-driver
tree.  It's useless for building in the kernel.  Let's move a few
lines in it to sound/core.h and remove it.
With this patch, sound/driver.h isn't removed but has just a single
compile warning to include it.  This should be really killed in
future.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Remove PCM sleep_min and tick
Takashi Iwai [Tue, 8 Jan 2008 17:09:57 +0000 (18:09 +0100)]
[ALSA] Remove PCM sleep_min and tick

The 'tick' in PCM is set (again) via sw_params.  And, nobody uses
this feature at all except for a command line option of aplay.
(This is literally 'nobody', as I checked alsa-lib API calls in all
 programs in major distros.)
Above all, if we need finer wake-ups for the position update, it's
basically an issue that the driver should solve, not tuned by each
application.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] PCM - clean up snd_pcm_lib_read/write
Takashi Iwai [Tue, 8 Jan 2008 17:08:14 +0000 (18:08 +0100)]
[ALSA] PCM - clean up snd_pcm_lib_read/write

Introduce a common helper function for snd_pcm_lib_read and snd_pcm_lib_write
for cleaning up the code.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Remove PCM xfer_align sw params
Takashi Iwai [Tue, 8 Jan 2008 17:05:26 +0000 (18:05 +0100)]
[ALSA] Remove PCM xfer_align sw params

The xfer_align sw_params parameter has never been used in a sane manner,
and no one understands what this does exactly.  The current
implementation looks also buggy because it allows write of shorter size
than xfer_align.  So, if you do partial writes, the write isn't actually
aligned at all.
Removing this parameter will make some pcm_lib_* code more readable
(and less buggy).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Fix PCM write blocking
Takashi Iwai [Tue, 8 Jan 2008 17:00:04 +0000 (18:00 +0100)]
[ALSA] Fix PCM write blocking

The snd_pcm_lib_write1() may block in some weird condition:
  - the stream isn't started
  - avail_min is big (e.g. period size)
  - partial write up to buffer_size - avail_min
The patch fixes this invalid blocking problem.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Remove indirect control access
Takashi Iwai [Tue, 8 Jan 2008 16:57:26 +0000 (17:57 +0100)]
[ALSA] Remove indirect control access

This patch removes the indirect control access to the control elements.
The indirect access has never been used and is even broken on 32bit
ioctl wrapper.  Let's clean it up.
The pointers still remain in snd_ctl_elem_* structs just to make sure
that the struct size won't change.  Once after checking the size
consistency, we can get rid of them, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Add model=laptop for HP 350 laptop
Takashi Iwai [Tue, 8 Jan 2008 16:19:22 +0000 (17:19 +0100)]
[ALSA] hda-codec - Add model=laptop for HP 350 laptop

Added the proper model=laptop for HP 350 laptop with Cxt5045 codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Add test model for ALC268
Jonathan Woithe [Tue, 8 Jan 2008 11:33:19 +0000 (12:33 +0100)]
[ALSA] hda-codec - Add test model for ALC268

This implements a test model for the ALC268.  It depends on the feature
added by alc260-test-eapd-0.2.diff.  This patch also adds a mention of
the ALC260 test model to ALSA-Configuration.txt since this seems to have
been missed.

Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Add EAPD controls for ALC260 test model
Jonathan Woithe [Tue, 8 Jan 2008 11:16:54 +0000 (12:16 +0100)]
[ALSA] hda-codec - Add EAPD controls for ALC260 test model

This implements a switch control for the EAPD signal output by the ALC26x
chips.  Since some laptops may utilise this to activate useful things it
is handy to have a control for this in the ALC26x test models.  The patch
includes the control in the ALC260 test model.

Signed-off-by: Jonathan Woithe <jwoithe@physics.adelaide.edu.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] PCM core - remove SNDRV_PCM_TSTAMP_MMAP condition in snd_pcm_status()
Jaroslav Kysela [Tue, 8 Jan 2008 11:24:01 +0000 (12:24 +0100)]
[ALSA] PCM core - remove SNDRV_PCM_TSTAMP_MMAP condition in snd_pcm_status()

The condition caused that the returned ring buffer position does not match
with timestamp when SNDRV_PCM_TSTAMP_MMAP mode was enabled. Removing
condition makes unified behaviour and interrupt based timestamp can be
accessed via PCM_IOCTL_SYNC_PTR or mmaped status area.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: Dynamically create digital gain mixers
Matthew Ranostay [Tue, 8 Jan 2008 11:10:50 +0000 (12:10 +0100)]
[ALSA] hda: Dynamically create digital gain mixers

Dynamically create digital gain mixers for dmics that have out-amp
support. Also some 92HD73xx's codecs don't have DMIC gains, so this also
prevents creating dead mixers.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-intel - Support multiple devices
Takashi Iwai [Mon, 7 Jan 2008 14:16:37 +0000 (15:16 +0100)]
[ALSA] hda-intel - Support multiple devices

It turned out that there can be multiple HD-audio devices on a single
machine (e.g. on-board audio and HDMI on graphic cards), so we need to
support multiple devices with snd-hda-intel driver.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound: Use time_before, time_before_eq, etc.
Julia Lawall [Mon, 7 Jan 2008 12:33:45 +0000 (13:33 +0100)]
[ALSA] sound: Use time_before, time_before_eq, etc.

The functions time_before, time_before_eq, time_after, and time_after_eq
are more robust for comparing jiffies against other values.
A simplified version of the semantic patch making this change is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@ change_compare_np @
expression E;
@@
(
- jiffies <= E
+ time_before_eq(jiffies,E)
|
- jiffies >= E
+ time_after_eq(jiffies,E)
|
- jiffies < E
+ time_before(jiffies,E)
|
- jiffies > E
+ time_after(jiffies,E)
)
@ include depends on change_compare_np @
@@
#include <linux/jiffies.h>
@ no_include depends on !include && change_compare_np @
@@
  #include <linux/...>
+ #include <linux/jiffies.h>
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] es18xx: Enable wavetable input from ESS chips
Krzysztof Helt [Mon, 7 Jan 2008 11:24:45 +0000 (12:24 +0100)]
[ALSA] es18xx: Enable wavetable input from ESS chips

This patch enables wavetable chips ES689/ES69X connected to
ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal
from the wavetable is active.
It has no effect if there is no ESS wavetable chip present.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Add IEC958 digital out support for Lenovo Thinkpads T61/X61
Jerone Young [Mon, 7 Jan 2008 11:22:18 +0000 (12:22 +0100)]
[ALSA] hda-codec - Add IEC958 digital out support for Lenovo Thinkpads T61/X61

This patch adds IEC958 digital out support for the AD1984 sound card.
This card can be found in Lenovo Thinkapds T61/X61. The digital out is
not located on the Thinkpad, but optional docking station (it's coxial
digital out). I've add this support as it is done the exact same way
for the AD1983 & AD1884.
I have tested this patch with my Lenovo Thinkpad T61 hooked up to a
docking station (that has the digital coxial) and then run to my home
theater reciever. Works like a charm :-)

Signed-off-by: Jerone Young <jerone@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: STAC927x VREF fix
Matthew Ranostay [Mon, 7 Jan 2008 11:18:28 +0000 (12:18 +0100)]
[ALSA] hda: STAC927x VREF fix

Some laptops incorrectly assume the front input jack as a line in
instead of a microphone in. Which in turn disables the voltage
reference, in which non-amplified input is not possible.  This patch
enables VREF80 for the input jack.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] oxygen: use uintptr_t in pointer casts
Clemens Ladisch [Fri, 4 Jan 2008 08:22:20 +0000 (09:22 +0100)]
[ALSA] oxygen: use uintptr_t in pointer casts

When we store the DMA channel number in the substream's private_data
pointer, use uintptr_t as an intermediate step when casting from/to
unsigned int to prevent the compiler from whining when the pointer size
is different.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound/usb/usbaudio.c: fix build with CONFIG_PM=n
Andrew Morton [Mon, 24 Dec 2007 13:40:56 +0000 (14:40 +0100)]
[ALSA] sound/usb/usbaudio.c: fix build with CONFIG_PM=n

sound/usb/usbaudio.c: In function 'usb_audio_suspend':
sound/usb/usbaudio.c:3674: error: implicit declaration of function 'snd_pcm_sus\pend_all'

Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Sort ad1986a cfg table
Takashi Iwai [Mon, 24 Dec 2007 13:36:09 +0000 (14:36 +0100)]
[ALSA] hda-codec - Sort ad1986a cfg table

Sort the ad1986a config table by PCI SSID (the last toshiba entry was
added wrongly).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] add Asus Xonar driver
Clemens Ladisch [Sun, 23 Dec 2007 18:52:08 +0000 (19:52 +0100)]
[ALSA] add Asus Xonar driver

Add the snd-virtuoso driver for the Asus Virtuoso 200 chip used on the
PCI and PCI-E models of the Xonar sound card.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] add CMI8788 driver
Clemens Ladisch [Sun, 23 Dec 2007 18:50:57 +0000 (19:50 +0100)]
[ALSA] add CMI8788 driver

Add the snd-oxygen driver for the C-Media CMI8788 (Oxygen) chip, used on
the Asound A-8788, AuzenTech X-Meridian, Bgears b-Enspirer,
Club3D Theatron DTS, HT-Omega Claro, Razer Barracuda AC-1,
Sondigo Inferno, and TempoTec HIFIER sound cards.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - alc268 input_mux should be a selector instead of mixer
Jiang Zhe [Thu, 20 Dec 2007 12:13:13 +0000 (13:13 +0100)]
[ALSA] hda-codec - alc268 input_mux should be a selector instead of mixer

According to the [0003659], the node 0x23,0x24 is a selector.
I checked the alc268 spec on the REALTEK website and it showed that they
were selectors indeed.
However, current code implement the alc268 input_mux in a mixer way.

Signed-off-by: Jiang Zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Device ID for Toshiba laptop which uses AD1986A
Jiang Zhe [Thu, 20 Dec 2007 12:01:28 +0000 (13:01 +0100)]
[ALSA] hda-codec - Device ID for Toshiba laptop which uses AD1986A

The model laptop-eapd get rid of the high-pitched noise.
(ALSA bug#3662)

Signed-off-by: Jiang Zhe <zhe.jiang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Fix capture mixers of ALC662 models
Herton Ronaldo Krzesinski [Wed, 19 Dec 2007 16:49:02 +0000 (17:49 +0100)]
[ALSA] hda-codec - Fix capture mixers of ALC662 models

The commit that added support for ASUS P701 eeepc also changed the
mixers of other ALC662 models, duplicating entries for the Capture
items, making them to not work anymore. This fixes it by removing
duplicated entries using where possible the common alc662_capture_mixer.
Also alc662_capture_mixer should use alc662* functions and not alc882
(I checked /proc/asound/card0/codec* on an eepc model and it's ok).

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] neo1973: ASoC include pathname fix
Harald Welte [Wed, 19 Dec 2007 14:37:49 +0000 (15:37 +0100)]
[ALSA] neo1973: ASoC include pathname fix

Fix s3c24xx include file path changes in asoc driver

Signed-off-by: Harald Welte <laforge@openmoko.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] usb-audio: add UR-80 PCM quirk
Clemens Ladisch [Wed, 19 Dec 2007 13:25:24 +0000 (14:25 +0100)]
[ALSA] usb-audio: add UR-80 PCM quirk

Add a quirk entry to handle Edirol UR-80 audio I/O.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h
Andrew Paprocki [Wed, 19 Dec 2007 11:13:44 +0000 (12:13 +0100)]
[ALSA] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h

Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to
AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added
AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK,
and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet
rev 1.3.

Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Fix invalid access to non-existing dmux on STAC
Takashi Iwai [Tue, 18 Dec 2007 17:05:52 +0000 (18:05 +0100)]
[ALSA] hda-codec - Fix invalid access to non-existing dmux on STAC

The digital mux on STAC codecs doesn't always exist although the
driver builds dmux enum mixer elements unconditionally.
Now the driver creates 'digital input source' mixer elements only
when dmux is available.
Also, the patch adds the missing dmux definition for STAC925x.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] cs4270: wrong sample rate when CONFIG_SND_SOC_CS4270_VD33_ERRATA is set
Timur Tabi [Tue, 18 Dec 2007 14:42:53 +0000 (15:42 +0100)]
[ALSA] cs4270: wrong sample rate when CONFIG_SND_SOC_CS4270_VD33_ERRATA is set

When CONFIG_SND_SOC_CS4270_VD33_ERRATA is set, there was a mismatch between
the mclk_ratios[] and cs4270_mode_ratios[] arrays.  The two arrays have been
merged and code has been shuffled.  One side effect is that the
cs4270_set_dai_sysclk() and cs4270_set_dai_fmt() functions are available only
if I2C has been enabled.

Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] include/sound/: Spelling fixes
Joe Perches [Tue, 18 Dec 2007 12:14:21 +0000 (13:14 +0100)]
[ALSA] include/sound/: Spelling fixes

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound/: Spelling fixes
Joe Perches [Tue, 18 Dec 2007 12:13:47 +0000 (13:13 +0100)]
[ALSA] sound/: Spelling fixes

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] at73c213: replace spinlock in mixer functions with a mutex
Hans-Christian Egtvedt [Mon, 17 Dec 2007 16:30:06 +0000 (17:30 +0100)]
[ALSA] at73c213: replace spinlock in mixer functions with a mutex

This patch fixes the locking bug in the at73c213 SPI sound driver. This bug was
triggered because spinlocks were wrapped around the spi_sync call which might
sleep. The fix was to add a mutex to the sound driver and replace the spinlocks
in the mixer functions with mutex lock/unlock.
Tested on STK1000/STK1002.

Signed-off-by: Hans-Christian Egtvedt <hcegtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - sort pci quirk list
Takashi Iwai [Mon, 17 Dec 2007 16:14:18 +0000 (17:14 +0100)]
[ALSA] hda-codec - sort pci quirk list

Sort pci quirk list in the order of PCI SSID.
This makes easier to find out the buggy duplicated entries.
Thanks to Andy Shevchenko for providing the sort script.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Add missing device link
Takashi Iwai [Mon, 17 Dec 2007 15:24:04 +0000 (16:24 +0100)]
[ALSA] Add missing device link

Added the missing link to struct device from the card instance.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Avoid overload of PCM volume on Cx5045 codec
Takashi Iwai [Mon, 17 Dec 2007 13:32:49 +0000 (14:32 +0100)]
[ALSA] hda-codec - Avoid overload of PCM volume on Cx5045 codec

The PCM volume of Cx5045 codec has overload that isn't useful but
rather harmful.  Add a hack to override the amp info to set the max
level 0 dB.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda: STAC927x DMIC Cleanup
Matthew Ranostay [Mon, 17 Dec 2007 10:58:13 +0000 (11:58 +0100)]
[ALSA] hda: STAC927x DMIC Cleanup

Cleaned up STAC927x and added several subsystem id's for more laptops.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] hda-codec - Fix definition of AC_KNBCAP_DELTA to match spec
Andrew Paprocki [Mon, 17 Dec 2007 10:49:44 +0000 (11:49 +0100)]
[ALSA] hda-codec - Fix definition of AC_KNBCAP_DELTA to match spec

AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the Intel
HDA spec, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask.

Signed-off-by: Andrew Paprocki <andrew@ishiboo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] Add SNDRV_PCM_IOCTL_TSTAMP back to compat ioctl
Takashi Iwai [Mon, 17 Dec 2007 10:44:25 +0000 (11:44 +0100)]
[ALSA] Add SNDRV_PCM_IOCTL_TSTAMP back to compat ioctl

The replaced one should be re-added for older alsa-lib.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] PCM - added back TSTAMP ioctl for PCM (for old alsa-lib binaries)
Jaroslav Kysela [Mon, 17 Dec 2007 08:02:22 +0000 (09:02 +0100)]
[ALSA] PCM - added back TSTAMP ioctl for PCM (for old alsa-lib binaries)

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] usb audio suspend support
Oliver Neukum [Fri, 14 Dec 2007 13:42:41 +0000 (14:42 +0100)]
[ALSA] usb audio suspend support

This patch implements suspend/resume support for USB audio devices.
It works with the microphone in my camera.

Signed-off-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sis7019: support the SiS 7019 Audio Accelerator
David Dillow [Fri, 14 Dec 2007 13:40:23 +0000 (14:40 +0100)]
[ALSA] sis7019: support the SiS 7019 Audio Accelerator

Basic audio support for the SiS 7019 Audio Accelerator as found in the
SiS 55x SoC. There is currently no synth support at the moment, but
audio playback and capture with two periods per buffer has seen
extensive use. Arbitrary period and buffer sizes (with multiple periods
per buffer) have seen light testing, but are believed to be production
ready.

Signed-off-by: David Dillow <dave@thedillows.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] sound/core/seq: move declarations of globally visible variables to proper...
Marcin Ślusarz [Fri, 14 Dec 2007 11:59:50 +0000 (12:59 +0100)]
[ALSA] sound/core/seq: move declarations of globally visible variables to proper headers

sound/core/seq: move declarations of globally visible variables to proper headers

Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
16 years ago[ALSA] info_oss: move prototype of snd_card_info_read_oss to info.h
Marcin Ślusarz [Fri, 14 Dec 2007 11:58:45 +0000 (12:58 +0100)]
[ALSA] info_oss: move prototype of snd_card_info_read_oss to info.h

info_oss: move prototype of snd_card_info_read_oss to info.h

Signed-off-by: Marcin Ślusarz <marcin.slusarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>