ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removal
The pop-removal specific values are configured for TWL4030 codec
for OMAP3EVM through this patch.
Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 29 Oct 2009 09:58:10 +0000 (11:58 +0200)]
ASoC: TWL4030: Change APLL powering sequence
It seams that certain part of the twl4030 codec needs the APLL
enabled before they are enabled.
Paths which has any digital processing needs need the APLL
enabled before they can function.
For example the vibra output will have some random data after
it is enabled and before the APLL also enabled.
If only analog components are in use (analog bypass), than it
seams, that the APLL does not need to be enabled. This lowers
the power consumption with around ~0.005A.
Adding DAPM_SUPPLY to the Digital playback route and also
to the capture route to enable and disable the APLL.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jari Vanhala [Thu, 29 Oct 2009 09:58:09 +0000 (11:58 +0200)]
ASoC: TWL4030: Vibra motor stop fix when it is driven with audio
This patch fixes vibrator playing incoherently, when driven
with audio. There is something wrong in switch 3 at
H-bridge and VIBRA_SET still affects PWM generator.
Slowest value fixes things.
Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Mack [Thu, 29 Oct 2009 01:24:32 +0000 (02:24 +0100)]
ASoC: CS4270: export de-emphasis filter as ALSA control
The CS4270 codec features an de-emphasis filter for compensation of
audio material filtered by an 50/15 uS algorithm. Not sure whether we
should always enable it for 44100Hz sampling frequency, but it should at
least be configurable by the user.
Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Wed, 28 Oct 2009 15:47:48 +0000 (15:47 +0000)]
ASoC: Minor SMDK64xx WM8580 cleanups
Fix up some comments, remove all enable_pin() calls (edge widgets
are all enabled by default) and mark the microphone as disabled by
default since it requires a resistor fit to connect it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Wed, 28 Oct 2009 08:57:04 +0000 (10:57 +0200)]
ASoC: TWL4030: Remove bypass tracking
Since ASoC core now handling the codec bias differently
there is no need to do the tracking of bypass switch states
anymore.
Handling of the common bit for analog loopbacks is done with
DAPM_SUPPLY for the bypass paths.
Now this bit is only enabled when there is a complete analog
bypass path, compared to the previous implementation, when the
global switch was enabled if there were any of the analog
bypass switch was on (regardless if there were complete path or
not)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Oct 2009 10:26:47 +0000 (13:26 +0300)]
ASoC: TWL4030: use the twl4030-codec.h for register descriptions
Remove the register descriptions from the twl4030.h file and use
the linux/mfd/twl4030-codec.h instead, which has the codec
related register descriptions also.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Oct 2009 10:26:46 +0000 (13:26 +0300)]
OMAP: Platform support for twl4030_codec MFD
Add needed platform data for the twl4030_codec MFD on boards,
where the audio part of the twl4030 codec is used.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 22 Oct 2009 10:26:45 +0000 (13:26 +0300)]
MFD: twl4030: add twl4030_codec MFD as a new child to the core
New MFD child to twl4030 MFD device.
Reason for the twl4030_codec MFD: the vibra control is actually in the codec
part of the twl4030. If both the vibra and the audio functionality is needed
from the twl4030 at the same time, than they need to control the codec power
and APLL at the same time without breaking the other driver.
Also these two has to be able to work without the need for the other driver.
This MFD device will be used by the drivers, which needs resources
from the twl4030 codec like audio and vibra.
The platform specific configuration data is passed along to the
child drivers (audio, vibra).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Samuel Ortiz <sameo@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Current implementation of those two functions found in
arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at
all, so they both end with BUG() on that machine. That results in
ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta.
The patch corrects the problem by not calling those two functions when run on
OMAP1 class based machines.
Created against linux-2.6.32-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Barry Song [Fri, 16 Oct 2009 10:13:38 +0000 (18:13 +0800)]
ASoC: Fix possible codec_dai->ops NULL pointer problems
Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves
then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai
if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc.
access the ops field in these DAIs, panic will happen.
Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi [Thu, 15 Oct 2009 06:03:56 +0000 (09:03 +0300)]
ASoC: Codec driver for Texas Instruments tlv320dac33 codec
Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.
TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.
The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.
Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).
b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Igor Grinberg [Wed, 14 Oct 2009 07:20:26 +0000 (09:20 +0200)]
ASoC: finally enable support for eXeda and CM-X300
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il> Signed-off-by: Mike Rapoport <mike@compulab.co.il> CC: Mark Brown <broonie@opensource.wolfsonmicro.com> CC: alsa-devel@alsa-project.org Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Tue, 13 Oct 2009 16:39:56 +0000 (17:39 +0100)]
ASoC: Remove snd_soc_suspend_device()
The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Logan Li [Wed, 14 Oct 2009 02:10:38 +0000 (10:10 +0800)]
ALSA: HDA VIA: Remove 48k sample rate limit for S/PDIF
48 kHz limit is for slightly better stability, and sample rates other
than 48k (like 96k/192k) are for better sound quality.
We choose better quality, so remove the 48k limit.
Signed-off-by: Logan Li <loganli@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 13 Oct 2009 14:07:59 +0000 (16:07 +0200)]
ALSA: hda - Allow all formats as default for Nvidia HDMI
In the commit f0613d5752d8f7d1d02e6d40947f38877fdf9c90
ALSA: hda - Add full rates/formats support for Nvidia HDMI
the flag LIMITIED_RATE_FMT_SUPPORT was set as default, as I forgot
to clear before commit.
After a reboot on an ARM1176 which amounts to a softreset, it has been
noted that the ALSA driver does not get registered and the probe fails
with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process
of reading from a register the SL1TxBusy bit is set indicating that the
transceiver is busy and remains so until the default timeout occurs.
Set the Power down register 0x26 to an arbitrary value as specified in
the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take
their default state.
Signed-off-by: Philby John <pjohn@in.mvista.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Sun, 11 Oct 2009 15:38:29 +0000 (17:38 +0200)]
ALSA: hda - Fix mute sound with STAC9227/9228 codecs
On FSC laptops, the sound gets muted gradually when the volume is chnaged.
This is due to the wrong volume-knob widget setup. The delta bit (bit 7)
shouldn't be set for these devices.
This patch adds a new quirk to set the value 0x7f to the widget 0x24
instead of 0xff.
Ben Dooks [Mon, 12 Oct 2009 20:17:09 +0000 (21:17 +0100)]
ASoC: S3C: Remove <plat/audio.h>
Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h
as it provides nothing to the current kernel and is not in any future plans
for the system.
Signed-off-by: Ben Dooks <ben@simtec.co.uk> Signed-off-by: Simtec Linux Team <linux@simtec.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Eero Nurkkala [Mon, 12 Oct 2009 05:41:59 +0000 (08:41 +0300)]
ASoC: Serialize access to dapm_power_widgets()
Access to damp_power_widgets() is assumed to be single-threaded.
Concurrent accesses to dapm_power_widgets() may result in
unpredictable behavior.
Calls from:
close_delayed_work()
soc_codec_close()
soc_pcm_prepare()
soc_suspend()
soc_resume_deferred()
to snd_soc_dapm_stream_event() do not have the codec->mutex
taken to cover the call to dapm_power_widgets(). Thus, take
the mutex in these paths also to assure single-threaded use
of dapm_power_widgets().
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tobias Hansen [Mon, 12 Oct 2009 14:24:15 +0000 (16:24 +0200)]
ALSA: snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd
snd-usb-us122l: corrent error number for not probing US-144 on ehci-hcd
This is the correct error number for telling the USB system that this
driver is not for the device.
Wu Zhangjin [Sat, 10 Oct 2009 15:53:49 +0000 (23:53 +0800)]
ALSA: SND_CS5535AUDIO: Remove the X86 platform dependency
SND_CS5535AUDIO is available on Loongson(MIPS compatible) family
machines, and checked it with ARCH=x86_64, no relative compiling
warnings & errors, so, remove the platform dependency directly.
Stephen Rothwell [Mon, 12 Oct 2009 04:56:17 +0000 (15:56 +1100)]
sound: use semicolons to end statements
Fixes:
sound/pci/hda/patch_via.c: In function 'patch_vt1718S':
sound/pci/hda/patch_via.c:4951: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1716S':
sound/pci/hda/patch_via.c:5441: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt2002P':
sound/pci/hda/patch_via.c:5794: error: expected expression before 'return'
sound/pci/hda/patch_via.c: In function 'patch_vt1812':
sound/pci/hda/patch_via.c:6148: error: expected expression before 'return'
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: ice1724 - Make call to set hw params succeed on ESI Juli@
If two streams are started immediately after one another (such as a
playback and a recording stream), the call to set hw params fails with
EBUSY. This patch makes the call succeed, so playback and recording will
work properly.
Signed-off-by: David Henningsson <launchpad.web@epost.diwic.se> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Krzysztof Helt [Sun, 11 Oct 2009 10:48:00 +0000 (12:48 +0200)]
ALSA: snd_dma_pointer workaround for chipsets with buggy DMA
The chipsets with the isa_dma_bridge_buggy set do not stop DMA during
DMA counter reads. The DMA counter is read in two 8-bit read steps
on x86 platform. Sometimes, such reads happen during higher byte
change so the lower byte is already decremented (rolled over) but
the higher byte is not. It introduces an error that position is
moved 256 bytes ahead of the true position. Thus, the next DMA
position read can return a lower value then the previous read.
If the DMA position is decreased (reversed) the ALSA subsystem is
tricked into the playback underrun error and resets the playback.
It results in a "pop" during a playback.
Work around the issue by reading the counter twice and choosing a higher
value.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Sat, 10 Oct 2009 11:07:52 +0000 (19:07 +0800)]
ALSA: HDA VIA: Add smart5.1 function.
Smart 5.1 is for 3-jacks model, to reuse input pins as outputs.
While off, they act as "line out" / "line in" / "mic in".
While on, they acts as "line out" / "back left/right" / "center/lfe".
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Logan Li <loganli@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lydia Wang [Sat, 10 Oct 2009 11:07:43 +0000 (19:07 +0800)]
ALSA: HDA VIA: Change VT1708S & VT1702 hp mode controls
For VT1708S and VT1702, deactivate "Headphone Playback Volume" and
"Headphone Playback Mute" control if "Independent HP" mode is OFF.
and rename VT1702 "Independent HP" text.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn> Signed-off-by: Logan Li <loganli@viatech.com.cn> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Robert Hancock [Sat, 10 Oct 2009 04:08:58 +0000 (22:08 -0600)]
ALSA: ice1724: Fix surround on Chaintech AV-710
Fix the num_total_dacs setting for Chaintech AV710. The existing comment
that only PSDOUT0 is connected is correct, but since the card is using
packed AC97 mode to send 6 channels to the codec, num_total_dacs should be
set to 6 and not 2. This allows 6-channel surround to work. Also clarify
a comment regarding the additional WM8728 codec on this card (it's connected
to the SPDIF output and always receives the same data).
Signed-off-by: Robert Hancock <hancockrwd@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
projects / karo-tx-linux.git / log