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14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Tue, 3 Aug 2010 07:06:08 +0000 (09:06 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
Jerone Young [Tue, 3 Aug 2010 06:46:44 +0000 (01:46 -0500)]
ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed

Just as with the X301. The X300 does not have a way to do SPDIF either.
It does not have a dock connector, nor does it have the SPDIF through
the headphone jack.

This patch fixes it so X300 does not show SPDIF, since it cannot do it.

To add all Lenovo Thinkpads had different codec subsytem IDs:

X300:
http://launchpadlibrarian.net/34862838/Card0.Codecs.codec.0.txt

Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the...
Jerone Young [Tue, 3 Aug 2010 06:46:42 +0000 (01:46 -0500)]
ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF

The Lenovo X301 does not have the ability to connect to a docking
station to use the SPDIF port. It also does not have the ability to do
SPDIF though the headphone jack or Display Port jacks.

This patch fixes it so this is not exposed for the X301 and users do
think it has the ability to do SPDIF.

I tested both headphone & display port jacks and it is not there. I have
tested this patch and it works great.

Also to add the other Thinkpads have different subsystem codec IDs.
Here are examples:

X301:
http://launchpadlibrarian.net/31561902/Card0.Codecs.codec.0.txt

X200:
http://launchpadlibrarian.net/49055036/Card0.Codecs.codec.0.txt

W500:
http://launchpadlibrarian.net/36276057/Card0.Codecs.codec.0.txt

Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Mon, 2 Aug 2010 12:51:13 +0000 (14:51 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Do not try to create speaker NIDs for ALC268 if there aren't any
David Henningsson [Mon, 2 Aug 2010 11:13:25 +0000 (13:13 +0200)]
ALSA: hda - Do not try to create speaker NIDs for ALC268 if there aren't any

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/asoc' into for-next
Takashi Iwai [Mon, 2 Aug 2010 10:11:03 +0000 (12:11 +0200)]
Merge branch 'topic/asoc' into for-next

14 years agoMerge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc...
Takashi Iwai [Mon, 2 Aug 2010 10:10:52 +0000 (12:10 +0200)]
Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc

14 years agoASoC: omap-mcbsp: Remove period size constraint in THRESHOLD mode
Peter Ujfalusi [Thu, 29 Jul 2010 06:51:28 +0000 (09:51 +0300)]
ASoC: omap-mcbsp: Remove period size constraint in THRESHOLD mode

The use of sDMA packet mode in THRESHOLD mode removes the restriction on the
period size.
With the extended THRESHOLD mode user space can ask for any
period size it wishes, and the driver will configure the
sDMA and McBSP FIFO accordingly.

Replace the hw_rule for the period size with static constraint,
which will make sure that the period size will be always
even (to avoid prime period size, which could be possible in
mono stream)

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoASoC: omap-mcbsp: Support for sDMA packet mode
Peter Ujfalusi [Thu, 29 Jul 2010 06:51:27 +0000 (09:51 +0300)]
ASoC: omap-mcbsp: Support for sDMA packet mode

Utilize the sDMA controller's packet syncronization mode, when
the McBSP FIFO is in use (by extending the THRESHOLD mode).
When the sDMA is configured for packet mode, the sDMA frame size
does not need to match with the McBSP threshold configuration.
Uppon DMA request the sDMA will transfer packet size number of
words, and still trigger interrupt on frame boundary.

The patch extends the original THRESHOLD mode by doing the
following:

if (period_words <= max_threshold)
Current THRESHOLD mode configuration

Otherwise (period_words > max_threshold)
McBSP threshold = sDMA packet size
sDMA frame size = period size

With the extended THRESHOLD mode we can remove the constraint
for the maximum period size, since if the period size is
bigger than the maximum allowed threshold, than the driver
will switch to packet mode, and picks the best (biggest)
threshold value, which can divide evenly the period size.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoASoC: omap-mcbsp: Code cleanup in omap_mcbsp_dai_hw_params
Peter Ujfalusi [Thu, 29 Jul 2010 06:51:25 +0000 (09:51 +0300)]
ASoC: omap-mcbsp: Code cleanup in omap_mcbsp_dai_hw_params

To make the code a bit more readable, change the indexed
references to the omap_mcbsp_dai_dma_params elements with
pointer.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoASoC: omap-mcbsp: Restructure the code within omap_mcbsp_dai_hw_params
Peter Ujfalusi [Thu, 29 Jul 2010 06:51:26 +0000 (09:51 +0300)]
ASoC: omap-mcbsp: Restructure the code within omap_mcbsp_dai_hw_params

In preparation for the extended threshold mode (sDMA packet mode
support), the code need to be restructured.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Mon, 2 Aug 2010 07:13:14 +0000 (09:13 +0200)]
Merge branch 'topic/misc' into for-next

14 years agoALSA: usb - Correct audio problem for Hauppage HVR-850 and others rel. to urb data...
John S Gruber [Sun, 1 Aug 2010 13:53:37 +0000 (09:53 -0400)]
ALSA: usb - Correct audio problem for Hauppage HVR-850 and others rel. to urb data align

Match usb ids in usb/quirks-table.h for some Hauppage HVR-950Q models
and for the HVR850 model to those ids at the end of au0828-cards.c

Thanks to nhJm449 for pointing out the problem.

Signed-off-by: John S Gruber <JohnSGruber@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Fri, 30 Jul 2010 12:12:03 +0000 (14:12 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Add a warning for ignored pins with ALC259/268/269
Takashi Iwai [Fri, 30 Jul 2010 12:10:43 +0000 (14:10 +0200)]
ALSA: hda - Add a warning for ignored pins with ALC259/268/269

The current ALC259/268/269 parser ignores some pins as unhandled,
but user won't notice what goes wrong.  So, added a warning message
for the ignored pins as a hint.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Handle pin NID 0x1a on ALC259/269
Takashi Iwai [Fri, 30 Jul 2010 12:08:25 +0000 (14:08 +0200)]
ALSA: hda - Handle pin NID 0x1a on ALC259/269

The pin NID 0x1a should be handled as well as NID 0x1b.
Also added comments.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Fri, 30 Jul 2010 09:35:41 +0000 (11:35 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Shut up pins at power-saving mode with Conexnat codecs
Takashi Iwai [Fri, 30 Jul 2010 09:28:02 +0000 (11:28 +0200)]
ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs

Call snd_hda_shutup_pins() for power-saving and reboot-notifier in
patch_conexant.c as well as other codecs.  This will reduce the pop
noise in power-save mode.

Reference: bnc#624896
https://bugzilla.novell.com/show_bug.cgi?id=624896

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Fri, 30 Jul 2010 08:57:34 +0000 (10:57 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Prefer VREF50 if BIOS sets for Realtek codecs
Takashi Iwai [Fri, 30 Jul 2010 08:55:44 +0000 (10:55 +0200)]
ALSA: hda - Prefer VREF50 if BIOS sets for Realtek codecs

If BIOS sets up the input pin as VREF 50, use the value as is instead of
overriding forcibly to VREF 80.  This fixes the quality of inputs on
some devices like Packard-Bell M5210.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Fri, 30 Jul 2010 08:52:51 +0000 (10:52 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Handle missing NID 0x1b on ALC259 codec
Takashi Iwai [Fri, 30 Jul 2010 08:51:10 +0000 (10:51 +0200)]
ALSA: hda - Handle missing NID 0x1b on ALC259 codec

Since ALC259/269 use the same parser of ALC268, the pin 0x1b was ignored
as an invalid widget.  Just add this NID to handle properly.
This will add the missing mixer controls for some devices.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Fri, 30 Jul 2010 08:50:14 +0000 (10:50 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Share digital I/O parser in patch_realtek.c
Takashi Iwai [Fri, 30 Jul 2010 08:48:14 +0000 (10:48 +0200)]
ALSA: hda - Share digital I/O parser in patch_realtek.c

Make a helper function to parse the digital I/Os of all Realtek codecs
to simplify the code and to ensure the setups.
Also, initialize digital I/O pins properly in init callbacks.  Some BIOS
seem to leave pins uninitialized.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Increase the connection list size for ALC662
Takashi Iwai [Fri, 30 Jul 2010 08:37:29 +0000 (10:37 +0200)]
ALSA: hda - Increase the connection list size for ALC662

Some ALC662-compatible codecs like ALC892 may have more than 4
connections for the input source.  Use HDA_MAX_CONNECTIONS instead of
the fixed magic number 4.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Make error messages more verbose
Takashi Iwai [Fri, 30 Jul 2010 08:36:29 +0000 (10:36 +0200)]
ALSA: hda - Make error messages more verbose

Add a prefix and more information for error messages regarding the
connection-list in hda_codec.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/asoc' into for-next
Takashi Iwai [Thu, 29 Jul 2010 18:06:15 +0000 (20:06 +0200)]
Merge branch 'topic/asoc' into for-next

14 years agoASoC: fsi: Add new funtion for SPDIF
Kuninori Morimoto [Thu, 29 Jul 2010 07:48:32 +0000 (16:48 +0900)]
ASoC: fsi: Add new funtion for SPDIF

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: fsi: remove device id check
Kuninori Morimoto [Wed, 28 Jul 2010 02:57:45 +0000 (11:57 +0900)]
ASoC: fsi: remove device id check

Current FSI driver id is not only 0

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: fsi: remove unnecessary clock processing
Kuninori Morimoto [Wed, 28 Jul 2010 02:57:36 +0000 (11:57 +0900)]
ASoC: fsi: remove unnecessary clock processing

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Thu, 29 Jul 2010 13:37:43 +0000 (15:37 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Rename iMic to Int Mic on Lenovo NB0763
David Henningsson [Thu, 29 Jul 2010 12:46:42 +0000 (14:46 +0200)]
ALSA: hda - Rename iMic to Int Mic on Lenovo NB0763

The non-standard name "iMic" makes PulseAudio ignore the microphone.
BugLink: https://launchpad.net/bugs/605101
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/hda' into for-next
Takashi Iwai [Thu, 29 Jul 2010 13:32:40 +0000 (15:32 +0200)]
Merge branch 'fix/hda' into for-next

14 years agoMerge branch 'fix/hda' into topic/hda
Takashi Iwai [Thu, 29 Jul 2010 13:32:34 +0000 (15:32 +0200)]
Merge branch 'fix/hda' into topic/hda

14 years agoALSA: hda - Add a PC-beep workaround for ASUS P5-V
Takashi Iwai [Thu, 29 Jul 2010 13:30:02 +0000 (15:30 +0200)]
ALSA: hda - Add a PC-beep workaround for ASUS P5-V

ASUS P5-V provides a SSID that unexpectedly matches with the value
compilant with Realtek's specification.  Thus the driver interprets
it badly, resulting in non-working PC beep.

This patch adds a white-list for such a case; a white-list of known
devices with working PC beep.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Thu, 29 Jul 2010 11:49:53 +0000 (13:49 +0200)]
Merge branch 'topic/misc' into for-next

14 years agosound: oss: msnd: check request_region() return value
Kulikov Vasiliy [Thu, 29 Jul 2010 10:45:50 +0000 (14:45 +0400)]
sound: oss: msnd: check request_region() return value

request_region() may fail, if so return -EBUSY.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: msnd: check request_region() return value
Kulikov Vasiliy [Thu, 29 Jul 2010 10:45:24 +0000 (14:45 +0400)]
ALSA: msnd: check request_region() return value

request_region() may fail, if so return -EBUSY.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: asihpi: check return value of get_user()
Kulikov Vasiliy [Wed, 28 Jul 2010 16:41:56 +0000 (20:41 +0400)]
ALSA: asihpi: check return value of get_user()

get_user() may fail, if so return -EFAULT.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agosound: oss: midi_synth: check get_user() return value
Kulikov Vasiliy [Wed, 28 Jul 2010 16:41:17 +0000 (20:41 +0400)]
sound: oss: midi_synth: check get_user() return value

get_user() may fail, if so return -EFAULT.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: sb: check get_user() return value
Kulikov Vasiliy [Wed, 28 Jul 2010 16:40:51 +0000 (20:40 +0400)]
ALSA: sb: check get_user() return value

get_user() may fail, if so return -EFAULT.

[Fixed one missing place by tiwai]

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: tlv320dac33: Add support for automatic FIFO configuration
Peter Ujfalusi [Wed, 28 Jul 2010 12:26:55 +0000 (15:26 +0300)]
ASoC: tlv320dac33: Add support for automatic FIFO configuration

Platform parameter to enable automatic FIFO configuration when
the codec is in Mode1 or Mode7 FIFO mode.
When this mode is selected, the controls for changing
nSample (in Mode1), and UTHR (in Mode7) are not added.
The driver configures the FIFO configuration based on
the stream's period size in a way, that every burst will
read period size of data from the host.
In Mode7 we need to use a formula, which gives close enough
aproximation for the burst length from the host point
of view.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoASoC: tlv320dac33: Revisit the FIFO Mode1 handling
Peter Ujfalusi [Wed, 28 Jul 2010 12:26:54 +0000 (15:26 +0300)]
ASoC: tlv320dac33: Revisit the FIFO Mode1 handling

Replace the hardwired latency definition with platform data
parameter, and simplify the nSample parameter calculation.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoMerge branch 'fix/hda' into for-next
Takashi Iwai [Wed, 28 Jul 2010 15:45:11 +0000 (17:45 +0200)]
Merge branch 'fix/hda' into for-next

14 years agoALSA: hda - Assume PC-beep as default for Realtek
Takashi Iwai [Wed, 28 Jul 2010 15:43:36 +0000 (17:43 +0200)]
ALSA: hda - Assume PC-beep as default for Realtek

Enable PC-beep as default for hardwares that aren't compliant with the
SSID value Realtek requires.  In such a case, better to enable the beep
to avoid a regression.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: hda - Don't register beep input device when no beep is available
Takashi Iwai [Wed, 28 Jul 2010 15:37:16 +0000 (17:37 +0200)]
ALSA: hda - Don't register beep input device when no beep is available

We check now the availability of PC beep and skip the build of beep
mixers, but the driver still registers the input device.  This should
be checked as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'fix/hda' into topic/hda
Takashi Iwai [Wed, 28 Jul 2010 12:26:47 +0000 (14:26 +0200)]
Merge branch 'fix/hda' into topic/hda

14 years agoMerge branch 'fix/hda' into for-next
Takashi Iwai [Wed, 28 Jul 2010 12:26:43 +0000 (14:26 +0200)]
Merge branch 'fix/hda' into for-next

14 years agoALSA: hda - Fix pin-detection of Nvidia HDMI
Takashi Iwai [Wed, 28 Jul 2010 12:21:55 +0000 (14:21 +0200)]
ALSA: hda - Fix pin-detection of Nvidia HDMI

The behavior of Nvidia HDMI codec regarding the pin-detection unsol events
is based on the old HD-audio spec, i.e. PD bit indicates only the update
and doesn't show the current state.  Since the current code assumes the
new behavior, the pin-detection doesn't work relialby with these h/w.

This patch adds a flag for indicating the old spec, and fixes the issue
by checking the pin-detection explicitly for such hardware.

Tested-by: Wei Ni <wni@nvidia.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Wed, 28 Jul 2010 09:53:42 +0000 (11:53 +0200)]
Merge branch 'topic/misc' into for-next

14 years agoALSA: hpimsgx: fix wrong sizeof
Axel Lin [Wed, 28 Jul 2010 08:58:42 +0000 (16:58 +0800)]
ALSA: hpimsgx: fix wrong sizeof

The correct size should be sizeof(gRESP_HPI_SUBSYS_FIND_ADAPTERS),
sizeof(&gRESP_HPI_SUBSYS_FIND_ADAPTERS) is incorrect.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoASoC: TWL4030: Capture route DAPM event fix
Peter Ujfalusi [Mon, 26 Jul 2010 06:59:15 +0000 (09:59 +0300)]
ASoC: TWL4030: Capture route DAPM event fix

There is no need to handle POST_PMU, POST_PMD event with
the Capture Route widget.
It is enough to handle POST_REG event, since that will come
when the user changes the routing, and we will switch the needed
bits in the registers.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Tue, 27 Jul 2010 08:21:30 +0000 (10:21 +0200)]
Merge branch 'topic/misc' into for-next

14 years agoMerge branch 'devel' of git.alsa-project.org:alsa-kernel into topic/misc
Takashi Iwai [Tue, 27 Jul 2010 08:16:04 +0000 (10:16 +0200)]
Merge branch 'devel' of git.alsa-project.org:alsa-kernel into topic/misc

14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Mon, 26 Jul 2010 15:02:53 +0000 (17:02 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Fix max amp cap calculation for IDT/STAC codecs
Takashi Iwai [Mon, 26 Jul 2010 15:00:15 +0000 (17:00 +0200)]
ALSA: hda - Fix max amp cap calculation for IDT/STAC codecs

The commit afbd9b8448f4b7d15673c6858012f384f18d28b8
    ALSA: hda - Limit the amp value to write
introduced a regression for codec setups with amp offsets like IDT/STAC
codecs.  The limit value should be a raw value without offset calculation.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Mon, 26 Jul 2010 08:33:51 +0000 (10:33 +0200)]
Merge branch 'topic/misc' into for-next

14 years agosound: oss: waveartist: simplify waveartist_sleep()
Kulikov Vasiliy [Mon, 26 Jul 2010 08:29:22 +0000 (12:29 +0400)]
sound: oss: waveartist: simplify waveartist_sleep()

waveartist_sleep() uses loop with schedule_timeout() to unconditionally
wait for msec. Use schedule_timeout_uninteruptible() instead.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agosound: oss: au1550_ac97: simplify au1550_delay()
Kulikov Vasiliy [Mon, 26 Jul 2010 08:28:32 +0000 (12:28 +0400)]
sound: oss: au1550_ac97: simplify au1550_delay()

au1550_delay() uses loop with schedule_timeout() to unconditionally wait
for msec. Use schedule_timeout_uninteruptible() instead.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Mon, 26 Jul 2010 08:28:22 +0000 (10:28 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Ensure codec patch files are checked for the correct codec ID
David Henningsson [Wed, 2 Jun 2010 14:56:41 +0000 (16:56 +0200)]
ALSA: hda - Ensure codec patch files are checked for the correct codec ID

Signed-off-by: David Henningsson <diwic@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/asoc' into for-next
Takashi Iwai [Sun, 25 Jul 2010 06:35:46 +0000 (08:35 +0200)]
Merge branch 'topic/asoc' into for-next

14 years agoASoC: da7210: Add HeadPhone Playback Volume control
Kuninori Morimoto [Wed, 21 Jul 2010 05:12:16 +0000 (14:12 +0900)]
ASoC: da7210: Add HeadPhone Playback Volume control

HeadPhone Playback Volume control register of DA7210 has
reserved area. This patch considered it as mute.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Wed, 21 Jul 2010 13:02:56 +0000 (15:02 +0200)]
Merge branch 'topic/misc' into for-next

14 years agosound/oss: Remove dead CONFIG_SOFTOSS*
Christian Dietrich [Wed, 21 Jul 2010 12:35:17 +0000 (14:35 +0200)]
sound/oss: Remove dead CONFIG_SOFTOSS*

CONFIG_SOFTOSS* doesn't exist in Kconfig or somewhere
else, therefore removing all references for it from the source code.

Signed-off-by: Christian Dietrich <qy03fugy@stud.informatik.uni-erlangen.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/asoc' into for-next
Takashi Iwai [Wed, 21 Jul 2010 13:01:48 +0000 (15:01 +0200)]
Merge branch 'topic/asoc' into for-next

14 years agoMerge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc...
Takashi Iwai [Wed, 21 Jul 2010 13:01:07 +0000 (15:01 +0200)]
Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc

14 years agoASoC: TWL4030: Add configurable delay after digimic enable
Peter Ujfalusi [Tue, 20 Jul 2010 12:49:09 +0000 (15:49 +0300)]
ASoC: TWL4030: Add configurable delay after digimic enable

When digital microphones are connected to twl, delay is
needed after enabling the digimic interface of the codec.
Add new parameter for the setup data, which can be used
to pass the apropriate delay in ms after the digimic
interface has been enabled.

Without certain delay (in certain HW configuration) the
beggining of the recorded sample contains a glitch, which
is generated by the digital microphones.

Delaying the micbias1, 2 (which is the bias for the digimic0
or 1) does not help, since the glitch is coming after
switching the digimic interface.

Reversing the micbias and digimic enable order does not
work either (in that case the wait need to be added after
the micbias enabled).

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoMerge branch 'topic/asoc' into for-next
Takashi Iwai [Tue, 20 Jul 2010 12:59:01 +0000 (14:59 +0200)]
Merge branch 'topic/asoc' into for-next

14 years agoMerge branch 'fix/asoc' into for-next
Takashi Iwai [Tue, 20 Jul 2010 12:10:02 +0000 (14:10 +0200)]
Merge branch 'fix/asoc' into for-next

14 years agoALSA: hda-intel - fix function_id rework (add missing bitmask)
Jaroslav Kysela [Tue, 20 Jul 2010 10:11:25 +0000 (12:11 +0200)]
ALSA: hda-intel - fix function_id rework (add missing bitmask)

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
14 years agoASoC: Unconditionally enable WM8994 AIF1ADC TDM mode
Mark Brown [Tue, 20 Jul 2010 09:13:14 +0000 (10:13 +0100)]
ASoC: Unconditionally enable WM8994 AIF1ADC TDM mode

AIF1ADC TDM mode has no effect other than causing the ADCDAT line to
be tristated rather than driven low on clock cycles where there is no
data to be transmitted. If the clock cycle is idle then there should
be no devices using the data so tristating should have no adverse
effects.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoASoC: davinci: let platform data define edma queue numbers
Sekhar Nori [Mon, 19 Jul 2010 07:01:16 +0000 (12:31 +0530)]
ASoC: davinci: let platform data define edma queue numbers

Currently the EDMA queue to be used by for servicing ASP through
internal RAM is fixed to EDMAQ_0 and that to service internal RAM
from external RAM is fixed to EDMAQ_1.

This may not be the desirable configuration on all platforms. For
example, on DM365, queue 0 has large fifo size and is more suitable
for video transfers. Having audio and video transfers on the same
queue may lead to starvation on audio side.

platform data as defined currently passes a queue number to the driver
but that remains unused inside the driver.

Fix this by defining one queue each for ASP and RAM transfers in the
platform data and using it inside the driver.

Since EDMAQ_0 maps to 0, thats the queue that will be used if
the asp queue number is not initialized. None of the platforms
currently utilize ping-pong transfers through internal RAM so that
functionality remains unchanged too.

This patch has been tested on DM644x and OMAP-L138 EVMs.

Signed-off-by: Sekhar Nori <nsekhar@ti.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC:Support Samsung SoC(S5P) in I2Sv2
Chanwoo Choi [Tue, 20 Jul 2010 05:28:30 +0000 (14:28 +0900)]
ASoC:Support Samsung SoC(S5P) in I2Sv2

This patch modify I2Sv2 driver to support Samsung SoC(S5PV210).

Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoMerge branch 'for-2.6.35' into for-2.6.36
Mark Brown [Tue, 20 Jul 2010 08:52:25 +0000 (09:52 +0100)]
Merge branch 'for-2.6.35' into for-2.6.36

14 years agoASoC: Select wm_hubs automatically for WM8994
Chanwoo Choi [Tue, 20 Jul 2010 05:28:33 +0000 (14:28 +0900)]
ASoC: Select wm_hubs automatically for WM8994

Otherwise all machine drivers need to do so.

Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com>
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: Implement WM8994 AIF1ADC2 paths
Mark Brown [Mon, 19 Jul 2010 13:02:34 +0000 (14:02 +0100)]
ASoC: Implement WM8994 AIF1ADC2 paths

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoASoC: Explicitly disable DC servo on WM hubs headphone powerdown
Mark Brown [Mon, 10 May 2010 20:06:14 +0000 (21:06 +0100)]
ASoC: Explicitly disable DC servo on WM hubs headphone powerdown

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoASoC: eukrea-tlv320: add support for cpuimx35sd
Eric Bénard [Mon, 19 Jul 2010 08:40:32 +0000 (10:40 +0200)]
ASoC: eukrea-tlv320: add support for cpuimx35sd

Signed-off-by: Eric Bénard <eric@eukrea.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoMerge branch 'topic/hda' into for-next
Takashi Iwai [Mon, 19 Jul 2010 16:47:56 +0000 (18:47 +0200)]
Merge branch 'topic/hda' into for-next

14 years agoALSA: hda - Add conexant quirk for AMD based Lenovo G series machines
Jerone Young [Mon, 19 Jul 2010 13:30:58 +0000 (08:30 -0500)]
ALSA: hda - Add conexant quirk for AMD based Lenovo G series machines

This is a follow on patch adds support for AMD based Lenovo G series
machines, such as the Lenovo G555.

Signed-off-by: Jerone Young <jerone.young@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Mon, 19 Jul 2010 15:59:53 +0000 (17:59 +0200)]
Merge branch 'topic/misc' into for-next

14 years agoALSA: riptide: check kzalloc() result
Kulikov Vasiliy [Fri, 16 Jul 2010 16:15:59 +0000 (20:15 +0400)]
ALSA: riptide: check kzalloc() result

If kzalloc() fails exit with -ENOMEM.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: echoaudio: check kmalloc() result
Kulikov Vasiliy [Fri, 16 Jul 2010 16:15:43 +0000 (20:15 +0400)]
ALSA: echoaudio: check kmalloc() result

If kmalloc() fails exit with -ENOMEM.

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Ack-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Mon, 19 Jul 2010 15:42:31 +0000 (17:42 +0200)]
Merge branch 'topic/misc' into for-next

14 years agoMerge branch 'devel' of git.alsa-project.org:alsa-kernel into topic/misc
Takashi Iwai [Mon, 19 Jul 2010 15:42:09 +0000 (17:42 +0200)]
Merge branch 'devel' of git.alsa-project.org:alsa-kernel into topic/misc

14 years agoALSA: pcm core - add a safe check to the silence filling function
Jaroslav Kysela [Mon, 19 Jul 2010 14:37:39 +0000 (16:37 +0200)]
ALSA: pcm core - add a safe check to the silence filling function

In situation when appl_ptr is far greater then hw_ptr, the hw_avail value
can be greater than buffer_size. Check for this.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
14 years agoALSA: hda-intel - do not mix audio and modem function IDs
Jaroslav Kysela [Mon, 19 Jul 2010 13:52:39 +0000 (15:52 +0200)]
ALSA: hda-intel - do not mix audio and modem function IDs

The function IDs are different for audio and modem. Do not mix them.
Also, show the unsolicited bit in the function_id register.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
14 years agoASoC: imx: check kzalloc() result and fix memory leak
Kulikov Vasiliy [Fri, 16 Jul 2010 16:16:54 +0000 (20:16 +0400)]
ASoC: imx: check kzalloc() result and fix memory leak

If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: imx: check kzalloc() result and fix memory leak
Kulikov Vasiliy [Fri, 16 Jul 2010 16:16:34 +0000 (20:16 +0400)]
ASoC: imx: check kzalloc() result and fix memory leak

If kzalloc() fails we must exit with -ENOMEM. Also we must free
allocated runtime->private_data on error as it would be lost on next
call to snd_imx_open().

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: davinci: check kzalloc() result (typo)
Kulikov Vasiliy [Fri, 16 Jul 2010 16:16:17 +0000 (20:16 +0400)]
ASoC: davinci: check kzalloc() result (typo)

The code checks 'davinci_vc' after kzalloc() and do not checks
'davinci_vcif_dev' that kzalloc() result is assigned to. It seems that
it is a typo (autocompletion?).

Signed-off-by: Kulikov Vasiliy <segooon@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoASoC: fsi: Add specified ID for soc-audio
Kuninori Morimoto [Fri, 16 Jul 2010 10:51:06 +0000 (19:51 +0900)]
ASoC: fsi: Add specified ID for soc-audio

Specified ID is necessary, when some codecs are used with FSI.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
14 years agoMerge branch 'for-2.6.35' into for-2.6.36
Mark Brown [Sat, 17 Jul 2010 18:45:43 +0000 (19:45 +0100)]
Merge branch 'for-2.6.35' into for-2.6.36

14 years agoASoC: Remove duplicate AUX definition from WM8776
Mark Brown [Sat, 17 Jul 2010 13:20:17 +0000 (14:20 +0100)]
ASoC: Remove duplicate AUX definition from WM8776

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: stable@kernel.org
14 years agoASoC: TWL6040: Correct widget handling for drivers
Jorge Eduardo Candelaria [Thu, 15 Jul 2010 16:38:01 +0000 (11:38 -0500)]
ASoC: TWL6040: Correct widget handling for drivers

In order to reduce pop-noise at powering up/down of the DACs and Drivers,
these components have to be handled in a specific sequence. Headset,
Handsfree, and Earphone drivers are now registered as PGA components to
ensure DACs are enabled first.

Also, add a delay to leave time for DACs to settle before
continuing power up/down sequence.

Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com>
Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Fri, 16 Jul 2010 09:35:06 +0000 (11:35 +0200)]
Merge branch 'topic/misc' into for-next

14 years agoALSA: asihpi - Avoid useless assignment of returned index values.
Eliot Blennerhassett [Fri, 16 Jul 2010 05:51:01 +0000 (17:51 +1200)]
ALSA: asihpi - Avoid useless assignment of returned index values.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: asihpi - Avoid using c99 uintX types.
Eliot Blennerhassett [Fri, 16 Jul 2010 05:51:00 +0000 (17:51 +1200)]
ALSA: asihpi - Avoid using c99 uintX types.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoALSA: asihpi - HPI version 4.04.01
Eliot Blennerhassett [Fri, 16 Jul 2010 05:50:59 +0000 (17:50 +1200)]
ALSA: asihpi - HPI version 4.04.01

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
14 years agoMerge branch 'topic/misc' into for-next
Takashi Iwai [Fri, 16 Jul 2010 06:37:25 +0000 (08:37 +0200)]
Merge branch 'topic/misc' into for-next