Peter Ujfalusi [Tue, 16 Feb 2010 11:23:16 +0000 (13:23 +0200)]
ASoC: tlv320dac33: Correct the OSCSET calculation
OSCSET calculation was not correct in case of 44.1KHz
sampling rate.
With small adjustment both 48 and 44.1 KHz calculation
now gives the correct value.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Tue, 16 Feb 2010 11:23:15 +0000 (13:23 +0200)]
ASoC: tlv320dac33: Clearing FIFOFLUSH flag before playback
In repeated playback the FIFOFLUSH bit remained set, and
never has been cleared.
Clear it during the setup phase.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 12 Feb 2010 11:05:44 +0000 (11:05 +0000)]
ASoC: Make pmdown_time a per-card setting
Make the pmdown_time a per-card setting rather than a global one,
initialised before the card initialisation runs. This allows cards
to override the default setting if it makes sense to do so (for
example, due to an unavoidable pop).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Takashi Iwai [Mon, 15 Feb 2010 16:05:28 +0000 (17:05 +0100)]
ALSA: hda - Correct ASUA blacklist for MSI brokenness
The MSI blacklist entry for ASUS mobo added in the commit 8ce28d6abff34886d3797b25324c940471b99164 was based on the alsa-info
output wrongly posted. Fix the id to the right one now.
Giuliano Pochini [Sun, 14 Feb 2010 17:16:10 +0000 (18:16 +0100)]
ALSA: Echoaudio - Add suspend support #2
This patch adds rearranges parts of the initialization code and adds
suspend and resume callbacks.
This patch adds suspend and resume callbacks.
It also rearranges parts of the initialization code so it can be
used in both the first initialization (when the module is loaded we
also have to load default settings) and the resume callback (where
we have to restore the previous settings).
Giuliano Pochini [Sun, 14 Feb 2010 17:15:59 +0000 (18:15 +0100)]
ALSA: Echoaudio - Add suspend support #1
Move the controls init code outside the init_hw() function because is must
not be called during resume.
This patch moves the code that initializes the card's controls with
default valued from the init_hw() function into a separated
set_mixer_defaults() function (one for each of the 16 supported
cards). This change is necessary because during resume we must
resurrect the hardware without losing the previous
settings. set_mixer_defaults() must be called only once when the
module is loaded.
Giuliano Pochini [Sun, 14 Feb 2010 17:15:51 +0000 (18:15 +0100)]
ALSA: Echoaudio - Add firmware cache #2
This patch implements a simple cache for the firmware files when CONFIG_PM is defined.
This patch changes get_firmware(), free_firmware() and adds
free_firmware_cache(). The first two functions implement a very
simple cache and the latter is used to actually release all the stored
firmwares when the module is unloaded.
When CONFIG_PM is not enabled those functions act as before, that is
free_firmware() releases the firmware immediately and
free_firmware_cache() does nothing.
Giuliano Pochini [Sun, 14 Feb 2010 17:15:34 +0000 (18:15 +0100)]
ALSA: Echoaudio - Add firmware cache #1
Changes the way the firmware is passed through functions.
When CONFIG_PM is enabled the firmware cannot be released because the
driver will need it again to resume the card.
With this patch the firmware is passed as an index of the struct
firmware card_fw[] in place of a pointer. That same index is then used
to locate the firmware in the firmware cache.
Takashi Iwai [Fri, 12 Feb 2010 17:17:06 +0000 (18:17 +0100)]
ALSA: hda - use WARN_ON_ONCE() for zero-division detection
Replace the zero-division warning message with WARN_ON_ONCE() per the
advice by Linus. This shouldn't happen, but if it happens, it's
possible that the bug happens often due to buggy IRQs.
Mark Brown [Thu, 11 Feb 2010 13:27:19 +0000 (13:27 +0000)]
ASoC: Add WM2000 driver
The WM2000 is a low power, high quality handset receiver speaker
driver with Wolfson myZone™ Ambient Noise Cancellation (ANC). It
provides enhanced voice communication quality in a noisy environment
if the handset acoustics are designed appropriately.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Thomas Weber [Thu, 11 Feb 2010 15:13:59 +0000 (16:13 +0100)]
Add ASoC support for Devkit8000
This patch expands the omap3beagle sound soc for the
beagle board clone DevKit8000.
Signed-off-by: Thomas Weber <weber@corscience.de> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jaroslav Kysela [Thu, 11 Feb 2010 16:50:44 +0000 (17:50 +0100)]
ALSA: usbmixer - add possibility to remap dB values
USB devices tends to represent dB ranges in different way than ALSA expects.
Add possibility to override these values and add guessed values for
SoundBlaster MP3+.
Also rename 'Capture Input Source' control to 'Capture Source' for
SoundBlaster MP3+ and Extigy.
Paul Menzel [Tue, 9 Feb 2010 10:42:27 +0000 (11:42 +0100)]
ASoC: Typo. s/Freecale/Freescale/
Signed-off-by: Paul Menzel <paulepanter@users.sourceforge.net> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Tue, 9 Feb 2010 13:24:04 +0000 (15:24 +0200)]
ASoC: TWL4030: Add supply for audio serial interface control
The serial interface (TDM/I2S) for the audio block have been
constantly enabled.
Introduce a new DAPM_SUPPLY for handling the AIF_EN bit, so
the interface is only enabled, when there is a need for it.
For example when only the analog loopback is enabled, there
is no need to keep the serial interface active.
I have added the persons who contributed to the Voice path
of twl4030 codec driver, so they might have the ability
to test this patch, and send an update for the Voice path,
if it is necessary
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Mack [Mon, 8 Feb 2010 18:32:59 +0000 (02:32 +0800)]
ASoC: cs4270: enable regulators at probe time
Enable the bulk regulators at probe time so we can safely disable them
again when going to suspend without confusing the reference counter.
Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jody Bruchon [Sat, 6 Feb 2010 15:46:26 +0000 (10:46 -0500)]
ALSA: hda-intel: Avoid divide by zero crash
On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by
zero
for as-yet unknown reasons. A simple check for zero prevents it, though
the problem that causes it remains. Since the workaround is harmless and
won't affect anyone except victims of this bug, it should be safe;
moreover,
because this crash can be triggered by a user-mode application, there are
denial of service implications on the systems affected by the bug without
the patch.
Takashi Iwai [Mon, 8 Feb 2010 14:16:08 +0000 (15:16 +0100)]
ALSA: hda - Merge HP mute-LED status callback on both IDT 92HD7x and 8x codecs
Merge the mute-LED status callback function for both IDT 92HD7x and 8x
codecs to one function. Also it's changed to check all DACs, and called
in the initialization to sync with the current status.
Takashi Iwai [Mon, 8 Feb 2010 14:06:13 +0000 (15:06 +0100)]
ALSA: hda - Detect HP mute-LED GPIO setup from GPIO counts
The GPIO pin number for the mute LED control on HP laptops can be
determined more easily by checking the number of available GPIO pins
of the codec chip. On a small package with up to 3 GPIOs, GPIO 0 is
used while GPIO 3 is used for others.
This fixes the missing mute GPIO for some HP laptops with new codecs.
Jody@Tritech [Sat, 6 Feb 2010 15:46:26 +0000 (10:46 -0500)]
ALSA: hda-intel: Avoid divide by zero crash
On my AMD780V chipset, hda_intel.c can crash the kernel with a divide by
zero
for as-yet unknown reasons. A simple check for zero prevents it, though
the problem that causes it remains. Since the workaround is harmless and
won't affect anyone except victims of this bug, it should be safe;
moreover,
because this crash can be triggered by a user-mode application, there are
denial of service implications on the systems affected by the bug without
the patch.
Jaroslav Kysela [Fri, 5 Feb 2010 09:19:41 +0000 (10:19 +0100)]
ALSA: ice1724 - aureon - fix wm8770 volume offset
The volume register is from 0..0x7f and 0..0x1a range is mute.
Also, fix mute combining in wm_vol_put(). The wrong behaviour was
noticed by Peter Christensen.
Maxim Levitsky [Thu, 4 Feb 2010 20:21:47 +0000 (22:21 +0200)]
ALSA: hda - Delay switching to polling mode if an interrupt was missing
My sound codec seems sometimes (very rarely) to omit interrupts (ALC268)
However, interrupt mode still works.
Thus if we get timeout, poll the codec once.
If we get 3 such polls in a row, then switch to polling mode.
This patch is maybe an bandaid, but this might be a workaround for hardware bug.
ALSA: ice1712: fix: lock samplerate when samplerate locking is enabled
I found that the sampling rate locking setting of the ice1712 sound driver
was only half-respected : when the driver was locked to, let's say, 44100Hz,
and a usermode app was requesting 48000Hz playback, the request was succesful
although the soundcard would continue to run at 44100Hz.
Here's a patch that will make those requests to fail.
Jaroslav Kysela [Tue, 2 Feb 2010 18:58:25 +0000 (19:58 +0100)]
ALSA: ctxfi - fix PTP address initialization
After hours of debugging, I finally found the reason why some source
and runtime combination does not work. The PTP (page table pages)
address must be aligned. I am not sure how much, but alignment to
PAGE_SIZE is sufficient. Also, use ALSA's page allocation routines
to ensure proper virtual -> physical address translation.
Cc: <stable@kernel.org> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Kailang Yang [Thu, 4 Feb 2010 13:16:14 +0000 (14:16 +0100)]
ALSA: hda - Add ALC269VB support
- Add new models ALC269VB_AMIC ALC269VB_DMIC
- Add alc269vb_laptop_dmic_setup
The record source index Dmic is 0x6 for ALC269VB.
- Change eeepc words for ALC269
- Modify init_verb tables of patch_alc269 patch_alc662 patch_alc882
- Modify common patch for ALC270 ALC269VB ALC275
Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Peter Ujfalusi [Thu, 4 Feb 2010 07:10:10 +0000 (09:10 +0200)]
ASoC: TWL4030: Module unloading fix
The module unloading path had several problems:
- it freed up the private structure twice
- it freed up the codec structure, which was allocated as part
of the private structure
- it did not freed up the reg_cache
- it did not unregistered the dais and the codec
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 3 Feb 2010 19:33:49 +0000 (19:33 +0000)]
ASoC: Add WM8912 DAC support
The WM8912 is a DAC only device register compatible with the WM8904
CODEC with ADC portions omitted. Support it within the WM8904 driver
based on the configured I2C device name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Wed, 3 Feb 2010 19:51:33 +0000 (19:51 +0000)]
ASoC: Optimise WM8904 output stage power control
Handle the output PGAs as part of the output powerup since they can
never be powered separately and reorder things so that we remove the
output shorts after both line and headphone outputs have been brought
up, minimising the opportunity for any issues.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Tue, 26 Jan 2010 22:37:11 +0000 (22:37 +0000)]
ASoC: Add support for BIAS_OFF when idle to WM8904
As well as disabling the biases of the CODEC the drop into BIAS_OFF will
also disable all the regulators powering the CODEC, allowing even greater
power savings on appropriately configured systems.
Since the regulator API does not currently provide notification when
regulators are disabled we assume that this always happens when we stop
using the regulators. Once 2.6.34 is merged this code can be optimised
to only sync the cache when power was actually removed.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Wed, 3 Feb 2010 17:55:55 +0000 (17:55 +0000)]
ASoC: Disable WM8993 regulators when turning bias off
While the regulators are disabled we cache all register writes.
Currently we assume that the regulator disable actually takes
effect, after the merge with the regulator tree in 2.6.34 the
regulator API will be able to notify us if the power is actually
removed (due to constraints or regulator sharing it may not be).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Mark Brown [Mon, 1 Feb 2010 18:48:03 +0000 (18:48 +0000)]
ASoC: Add a cache_sync bit to the CODEC structure
Add a bit to the CODEC structure indicating if a cache sync is required.
By default this will be set if a cache only write is done to a soc-cache
register cache. This allows us to avoid syncing the cache back after
using cache only writes if there were no changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Charles Chin [Thu, 4 Feb 2010 09:28:02 +0000 (10:28 +0100)]
ALSA: hda - Fix docking output for IDT 92HD8xx codecs
This patch fixes docking output support for IDT 92HD81/83/88 family codecs.
Typically one of ports 0xE or 0xF is used for docking output, while only
port 0xF is common on all the three codec families. We don't want the
pin to select the analog mixer here.
Signed-off-by: Charles Chin <Charles.Chin@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mark Brown [Mon, 1 Feb 2010 18:46:10 +0000 (18:46 +0000)]
ASoC: Allow CODECs to ask soc-cache to suppress physical writes
Currently the soc-cache code will always write to the device, meaning
that we need the device to be powered and active at pretty much all
times the system is active. Allowing cache only writes lays some
groundwork for future enhancements to allow devices to be put into a
full off state when the audio subsystem is idle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: fix compilation breakage in sound/soc/sh/fsi.c
ctrl_outl() has become void at some point, which breaks compilation of fsi.c.
Make writing functions void, as their output is anyway not evaluated, and use
__raw_writel and __raw_readl instead of deprecated ctrl_outl and ctrl_inl
respectively.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Mack [Tue, 2 Feb 2010 10:45:27 +0000 (18:45 +0800)]
ASoC: fix PXA SSP port resume
Unconditionally save the register states when suspending and restore
them again at resume time. Register contents were not preserved over
suspend, and hence the driver takes false assumptions about them.
The clock must be enabled to access the register block.
Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Joe Perches [Tue, 2 Feb 2010 07:22:16 +0000 (23:22 -0800)]
ASoC: Fix continuation line formats
String constants that are continued on subsequent lines with \
are not good.
Signed-off-by: Joe Perches <joe@perches.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Fri, 29 Jan 2010 17:47:12 +0000 (17:47 +0000)]
ASoC: Add WM8994 CODEC driver
The WM8994 is a highly integrated ultra-low power hi-fi audio subsystem
designed for smartphones and other portable devices rich in multimedia
features. It provides advanced digital mixing facilities enabling low
power high quality interconnection of CPU, baseband and other audio
sources through flexible digital and analogue routing, and integrates
a class W headphone driver and stereo class D speaker drivers.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 20 Jan 2010 17:39:45 +0000 (17:39 +0000)]
ASoC: Improved wm_hubs headphone handling
Perform DC servo offset calibration using a series update sequence
rather than startup update sequence, tuning the configuration of the
WM8993 DC servo to make best use of this.
Also introduce currently unused data allowing us to correct for
any systematic errors in the DC servo calibration results and an
alternative startup path for the headphone output which performs
better with some chip revisions. The alternative setup sequence is
enabled for WM8993.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Joe Perches [Sun, 31 Jan 2010 20:02:12 +0000 (12:02 -0800)]
ASoC: Fix continuation line formats
String constants that are continued on subsequent lines with \
are not good.
Signed-off-by: Joe Perches <joe@perches.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: improve MCLKDIV calculation in wm8978, when OPCLK is not used
In case, if OPCLK is not used, and PLL is used for driving the codec, the
choice of PLL output frequency could result in a needlessly imprecise
system clock frequency. Use an iterative process to select a precise
configuration.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>