Takashi Iwai [Tue, 28 Jun 2011 06:59:30 +0000 (08:59 +0200)]
ALSA: hda - Fix warnings with CONFIG_SND_POWER_SAVE=n
Use static inline for dummy function to fix the warnings like below
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_init’:
sound/pci/hda/patch_sigmatel.c:4387:3: warning: statement with no effect
sound/pci/hda/patch_sigmatel.c: In function ‘stac92xx_resume’:
sound/pci/hda/patch_sigmatel.c:4927:3: warning: statement with no effect
Takashi Iwai [Mon, 27 Jun 2011 14:17:07 +0000 (16:17 +0200)]
ALSA: hda - More volume-init fixes for ALC267 codec
More similar fixes like previous commits: handle the exceptional case
like ALC267 where no volume amp is found in ADC widget but in the
capsrc widget instead.
Also minor checks for avoiding possible erros: no connection-select
when the pin has a single selection, and add beep verbs only when the
0x1d is used for beep.
Takashi Iwai [Mon, 27 Jun 2011 13:53:38 +0000 (15:53 +0200)]
ALSA: hda - Fix volume-init for ALC259 with invalid widget caps
ALC259 seems to provide an invalid widget capability for the input-src
selector widget. The widget shows the input-amp while it's a selector,
and this confuses the current ALC882 initialization code that is used
for ALC259, too. For fixing this, check the amp capability and handle
the connection selection individually.
Also, ALC259 has no mute bit in DAC volume, so we need to initialize
it as ZERO instead of MUTE.
Takashi Iwai [Mon, 27 Jun 2011 13:48:17 +0000 (15:48 +0200)]
ALSA: hda - Fix volume-init of ALC299 & co
ALC269 and compatible codecs have the output volume in DACs, thus we
can't use the ALC880's code as is. Fixed by checking the amp caps and
picking up the right widget for initialization.
The input amps of mixer widgets should be unmuted as default (as
usually they have no assigned mixer switches).
More fixes in this commit are, however, for ALC260: ALC260 codec can
have multiple output mixers connnected to a single DAC althouh the
driver didn't pick up them properly.
Takashi Iwai [Mon, 27 Jun 2011 10:34:01 +0000 (12:34 +0200)]
ALSA: hda - Check hard-wired DACs at first for ALC662 & co
Some Realtek codecs have the output pins hardwired with certain DACs.
These DACs have to be assigned at first and assign the rest for
multi-DAC pins so that all DACs can be assigned properly.
Without such an optimization, speaker outputs may be assigned to the
same DAC as the headphone or others.
Takashi Iwai [Mon, 27 Jun 2011 09:32:07 +0000 (11:32 +0200)]
ALSA: hda - Call proper DAC-filler function for Realtek auto-parser
In alc_auto_add_multi_channel_mode(), when the primary HP workaround
is enabled, it re-initializes the DAC list but calls alc662's function
in a fixed way. This isn't pretty suitable for other codecs, of course.
Now we call it with fill_dac function pointer so that the proper
function can be called at that point.
Sangbeom Kim [Mon, 20 Jun 2011 08:29:29 +0000 (17:29 +0900)]
ASoC: SAMSUNG: Add WM8994 PCM Machine driver
This patch add WM8994 PCM machine driver to support PCM audio
on SMDKV310, SMDKC210 boards.
Playback and Capture supports 8kHz sampling rates.
and It is tested on SMDKV310, SMDKC210.
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com> Acked-by: Jassi Brar <jaswinder.singh@linaro.org> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Fri, 24 Jun 2011 12:10:28 +0000 (14:10 +0200)]
ALSA: hda - Add snd_hda_get_conn_list() helper function
Add a new helper function snd_hda_get_conn_list().
Unlike snd_hda_get_connections(), this function doesn't copy the
connection-list but gives the raw pointer for the cached list.
Tim Blechmann [Fri, 24 Jun 2011 15:36:20 +0000 (17:36 +0200)]
ALSA: lx6464es - include mac address in device name
each device has a unique mac address, which can be used to distinguish
multiple devices in the same machine. we therefore include the full mac
address in the device shortname and the last 6 bytes in the device id.
Signed-off-by: Tim Blechmann <tim@klingt.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Jesper Juhl [Thu, 23 Jun 2011 21:54:40 +0000 (23:54 +0200)]
ALSA: Remove unneeded version.h includes from sound/
In the sound/ directory there are two files (flagged by 'make
versioncheck'); sound/pci/asihpi/asihpi.c and
sound/soc/codecs/wm8991.c that include linux/version.h although they
don't need it. This patch removes the unneeded includes.
Takashi Iwai [Fri, 24 Jun 2011 09:03:58 +0000 (11:03 +0200)]
ALSA: hda - Clean up multi-channel mixer name assignment in patch_realtek.c
Change alc_get_line_out_pfx() in patch_realtek.c to provide the channel
specific name and assign the index so that each caller doesn't have to
set the channel name by itself.
Also, check the multi-io case with the primary hp-out; for the multi-io
channels, assign the channel name instead of "Headphone" with indices.
This makes the mixer names more intuitive and reduces confusion.
Takashi Iwai [Fri, 24 Jun 2011 08:43:03 +0000 (10:43 +0200)]
ALSA: hda - Add a workaround for invalid line-out setups
Some BIOS set up the pin config wrongly as line-out although it's
supposed to be a speaker out. In most cases, though, we can judge
the validity by checking the connection type -- when it's FIXED,
mostly it's an invalid line-out but a speaker.
Daniel Mack [Wed, 22 Jun 2011 18:48:25 +0000 (20:48 +0200)]
ASoC: pxa-ssp: Correct check for stream presence
Don't rely on the codec's channels_min information to decide wheter or
not allocate a substream's DMA buffer. Rather check if the substream
itself was allocated previously.
Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
Mark Brown [Wed, 22 Jun 2011 19:07:24 +0000 (20:07 +0100)]
ASoC: Add basic WM8918 support
The WM8918 is register compatible with the WM8904 with a subset of the
functionality. Add the device ID, a subsequent patch will ensure that only
the relevant functionality is exported to userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure] Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Johannes Stezenbach <js@sig21.net>
[zonque@gmail.com: transform to new ASoC structure] Signed-off-by: Daniel Mack <zonque@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Wed, 22 Jun 2011 13:23:25 +0000 (15:23 +0200)]
ALSA: hda - Implement dynamic-ADC switching for VIA codecs
Some VIA codecs like VT1702 provide the input-route only to specific
ADCs such as digital-mic inputs. These routes aren't covered by the
normal primary ADC, and for now, user had to open the capture stream
assigned to that special ADC manually for using such inputs.
This patch implements a way to switch the current ADC dynamically per
the input-source selection in such a case. When this workaround is
activated, the driver provides only one capture stream and one input-
source control but with the full possible inputs. The driver switches
the ADC to be used (or being used) according to the input-source on the
fly.
Taylor Hutt [Mon, 20 Jun 2011 18:54:32 +0000 (11:54 -0700)]
ASoC: codecs: Max98095: Fix logging of hardware revision.
The base hardware revision of the Maxim 98095 part is 0x40; the code
which outputs the revision of the hardware has been updated to
properly use uppercase alphabetic values for the revision numbers.
Also, the use of a constant for the length 'max98095_dai' has been
replaced with ARRAY_SIZE().
Signed-off-by: Taylor Hutt <thutt@chromium.org> Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Tue, 21 Jun 2011 12:22:14 +0000 (14:22 +0200)]
ALSA: hda - Fix surround-volume parsing for VT1708B codecs
The surround/CLFE/side DACs on VT1708B and co have no amp but the
connected selector widgets have the amp instead. Fix the parser to
check these selector widgets for the possible mixer controls as well.
Takashi Iwai [Tue, 21 Jun 2011 10:57:22 +0000 (12:57 +0200)]
ALSA: hda - Fix the check of loopback-mixer element index in patch_via.c
Fix the check of the multiple loopback-mixer, which gave sometimes
a wrong index assigned to an element even for different names, e.g.
Mic and Front Mic. Now check the label properly for avoid duplication.
Reported-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Tue, 21 Jun 2011 10:51:33 +0000 (12:51 +0200)]
ALSA: hda - Assign smart51 only in the same stack for VIA codecs
The input jacks assigned as the smart51 outputs must be in the same
stack, either rear, front or other. Also, prefer line-in as the surround
to mic-in.
Tony Vroon [Mon, 20 Jun 2011 21:11:11 +0000 (22:11 +0100)]
ALSA: hda - Remove ALC268 model override for CPR2000
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.
Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com> Signed-off-by: Tony Vroon <tony@linx.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 20 Jun 2011 10:39:26 +0000 (12:39 +0200)]
ALSA: hda - Initialize unsol events dynamically in patch_via.c
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs. Also, consolidate the unsol handlers for HP
and line-out mutes.
Takashi Iwai [Sun, 19 Jun 2011 14:24:21 +0000 (16:24 +0200)]
ALSA: hda - Fix independent-HP handling in patch_via.c
Fix races in handling of HP DAC and independent streams for VIA codecs.
Also, allow the HP output path without front-DAC, and removed
unnecessary activation of HP mixer elements.
This also removes the handling of shared side/HP stream; it's anyway
implemented in a broken way, so we need to re-implement the feature
later...
Takashi Iwai [Sat, 18 Jun 2011 14:40:14 +0000 (16:40 +0200)]
ALSA: hda - Assign HP-independent PCM to individual stream
Instead of using the secondary substream, create an individual PCM
stream for HP-independent PCM. Otherwise it's difficult to handle
different channel numbers with multi-channel stream in the sam PCM
stream structure.
Takashi Iwai [Fri, 17 Jun 2011 15:53:38 +0000 (17:53 +0200)]
ALSA: hda - Unify output-control parsing in patch_via.c
Parse the output-paths more dynamically, i.e. traverse the paths
from each output pin instead of fixed assignment for each codec.
Now all codecs are using the same output parser code.
The smart51 setup doesn't work with this change, and will be fixed
in the next commits.
Takashi Iwai [Fri, 17 Jun 2011 14:37:45 +0000 (16:37 +0200)]
ALSA: hda - Change pin-ctl for auto-muting in patch_via.c
Mute the outputs via pin-controls instead of amps for the auto-mute
handling. This makes our life easier as it avoids conflict of the states
between the mixer elements and the auto-mute toggles.
With this change, we can use vmaster for the master control easily now.
Takashi Iwai [Fri, 17 Jun 2011 13:46:13 +0000 (15:46 +0200)]
ALSA: hda - Defer mixer element creation to the right time in patch_via.c
The jack-detect control should be created at the time of build_controls
callback instead of calling snd_hda_add_ctls() at the tree-parsing time.
For that, copy the control to the temporary array like other cases.
Also, fixed typos of vt1708_jack_detect in all places.
Takashi Iwai [Fri, 17 Jun 2011 14:59:21 +0000 (16:59 +0200)]
ALSA: hda - Add control to suppress the dynamic pin-power for VIA
Currently VIA driver controls the power-state of each pin per jack
detection. But, it means that the power-state mismatch may occur when
the machine doesn't give the proper jack-detection.
For avoiding this problem, a new control element "Dynamic Power-Control"
is provided so that user can turn on/off the pin-power control.
Mark Brown [Sun, 19 Jun 2011 12:49:28 +0000 (13:49 +0100)]
ASoC: Fix DAPM sequence run for per-widget I/O methods
Previously we were using the DAPM context rather than a widget as the
argument for update_bits() so we didn't need to care that our list walk
of widgets left us one beyond the end of the list. Now we're using them
for the register update we need to make sure we're pointing at an actual
widget not the list_head.
Fix originally suggested by Liam on IM.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
Mark Brown [Sat, 18 Jun 2011 14:31:38 +0000 (15:31 +0100)]
ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency
The Sigma code is in drivers/firmware which is only included on a very
small subset of architectures and so ends up breaking the build on
others. There's a pending patch to make the directory build as standard
but it's not merged yet.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALSA: HDA: Remove redundant LPIB quirks for ATI chipset
Now that we have changed the position_fix default for ATI and AMD
to be LPIB (see commit 50e3bbf989), we can remove the quirks that
were added for ATI chipsets.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Fri, 17 Jun 2011 12:23:46 +0000 (14:23 +0200)]
ALSA: hda - Fix no NID error with VIA codecs
The via driver spews warnigs like
hda-codec: no NID for mapping control Independent HP:0:0
with some codecs because snd_hda_add_nid() is called with nid=0.
This patch fixes it by skipping the call when no corresponding widget
is found.
Mike Frysinger [Fri, 17 Jun 2011 02:24:15 +0000 (22:24 -0400)]
ASoC: Blackfin: allow SPI for SSM2602 parts
This board has hardware switches for selecting SPI or I2C, so don't
require I2C for this driver.
Signed-off-by: Mike Frysinger <vapier@gentoo.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clemens Ladisch [Fri, 17 Jun 2011 06:18:35 +0000 (08:18 +0200)]
ALSA: isight: adjust for new queueing API
Since commit 13882a82ee16 (optimize iso queueing by setting
wake only after the last packet), drivers are required to call
fw_iso_context_queue_flush() after queueing a batch of packets.
The missing call would have an effect only if the controller
queue underruns, but then the DMA would stop completely.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Fri, 17 Jun 2011 06:17:56 +0000 (08:17 +0200)]
ALSA: firewire-speakers, oxygen, ua101: allow > 10 s periods
Since commit f2b3614cefb6 (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Torsten Schenk [Thu, 16 Jun 2011 19:06:27 +0000 (21:06 +0200)]
ALSA: 6fire - Fix signedness bug
Fixed remaining issues of the signedness bug discovered by Dan Carpenter.
A check was remaining that tests if unsigned rt->rate is >= 0.
Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates)
as invalid rate value and not -1.
Barry Song [Wed, 15 Jun 2011 19:29:22 +0000 (15:29 -0400)]
ASoC: AD1836: rename suspend/resume funcs
Use less specific names for suspend/resume to match the probe/remove funcs
where these are now used.
Signed-off-by: Barry Song <barry.song@analog.com> Signed-off-by: Scott Jiang <scott.jiang@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mike Frysinger [Wed, 15 Jun 2011 19:29:19 +0000 (15:29 -0400)]
ASoC: AD1836: drop unnecessary spi register check
The only thing the init func does is register a spi driver, so if that
fails, we return the value back up to the caller who will display an
error message for us. So drop the redundant checking/message.
Signed-off-by: Mike Frysinger <vapier@gentoo.org> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>