Brian Bloniarz [Tue, 18 Jan 2011 07:20:03 +0000 (23:20 -0800)]
ALSA: ice1712 delta - initialize SPI clock
The driver was using an initial value for the clock on the SPI bus
which was read from ICE1712 EEPROM,
ice->eeprom.data[ICE_EEP1_GPIO_STATE] & ICE1712_DELTA_AP_CCLK (0x02)
It appears some cards have it default high, some cards
have it default low. On my Delta 66 rev. E:
$ cat /proc/asound/M66/ice1712 | grep 'GPIO state'
GPIO state : 0x70 /* ICE1712_DELTA_AP_CCLK bit is zero */
On my Audiophile 2496:
$ cat /proc/asound/M2496/ice1712 | grep 'GPIO state'
GPIO state : 0xfe /* ICE1712_DELTA_AP_CCLK bit is one */
It must be raised before the first SPI write happens, or the write will
fail, leading to:
[ 23.248721] invalid CS8427 signature 0x0: let me try again...
Raymond Yau [Mon, 17 Jan 2011 10:19:03 +0000 (11:19 +0100)]
ALSA: hda - Add add multi-streaming playback for AD1988
Attached a patch which add a new model to support multi-streaming
playback for ad1988.
playback another stereo stream through the front panel headphone on
device 2 while playback through the speakers connected to rear panel
on device 0 at the same time.
Tested with ad1988a rev2 codec on asus P5B-V motherboard.
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: EP93xx: fixed LRCLK rate and DMA oper. in I2S code
Changelog:
1. I2S module of EP93xx should be feed by 32bit DMA transfers. This is
hardware limitation and that's the way original Cirrus's driver worked.
This will fix distorted sound playback and make capture actually work in
present ep93xx drivers.
I've found, that author of code, on which modern ep93xx-i2s.c and
ep93xx-pcm.c are based, had faced this problem also in 2007:
http://blog.gmane.org/gmane.linux.ports.arm.cirrus/month=20070101/page=3
Now SoC code uses his developments, but not overcomes the hardware
issues. Some details from EP93xx users guide:
Both I2S transmitter and receiver have similar 16x32bit FIFO, where they
store 8 samples for both left and right channels. The FIFO is always
32bit wide and should be properly aligned if you use samples of other
width. Transmitter and receiver have configuration registers for
selection of I2S word length (16, 24, 32). They are I2STXWrdLen and
I2SRXWrdLen.
Yes, EP93xx DMA can do byte, word and quad-word transfers. The width for
transfers to and from peripherals is selected by particular module
configuration. Lucky AC97 module has such configuration: AC97RXCRx
registers, bit CM (Compact mode enable) switches between 16 and 32 bit
samples. AC97TXCRx registers have the same bits for transmitters.
ep93xx-ac97.c enables this compact mode and so has all the rights to use
S16_LE format.
No one has found such a configuration in I2S module until now in any
Cirrus manuals. I2S module always feeds it's 32bit wide FIFO with 32bit
samples consecutively for left and right channels. You cannot use 32-bit
DMA transfers to transfer two 16-bit samples.
So we can use two formats for AC97, but should remove all but S32_LE for
I2S. Always using 32 bit chunks is not a problem for I2S, the codec I
use uses less bits too (24), it's permitted by I2S standard.
In proposed patch formats list shortened to just S32_LE, this makes all
the DMA transactions right, while ALSA will do all sample format
translation for us.
2. Incorrect setting of LRCLK (2 times slower) in original ep93xx-i2s.c
masks the first problem.
DMA takes two 16 bit samples instead of one, overall sound speed seems
to be normal, but you get actually 4000 sampling rate instead of
requested 8000 and therefore some noise... This is also the reason why
the capture function not worked at all in this driver...
If we take a look into I2S specification, we will figure that LRCLK MUST
be equal to sample rate, if we are talking about stereo (in mono too,
but it's not our case at all).
In proposed patch SCLK and LRCLK rates are corrected, assuming we always
send 32 bits * 2 channels to codec.
Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Raymond Yau [Sun, 16 Jan 2011 02:55:54 +0000 (10:55 +0800)]
ALSA : au88x0 - Limit number of channels to fix Oops via OSS emu
Fix playback/capture channels patch to change supported playback
channels of au8830 to 1,2,4 and capture channels to 1,2.
This prevent oops when oss emulation use SNDCTL_DSP_CHANNELS to
set 3 Channels
Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASoC: WL1273 FM radio: Fix breakage with MFD API changes
These changes are needed to keep up with the changes in the
MFD core and V4L2 parts of the wl1273 FM radio driver.
Use function pointers instead of exported functions for I2C IO.
Also move all preprocessor constants from the wl1273.h to
include/linux/mfd/wl1273-core.h.
Signed-off-by: Matti J. Aaltonen <matti.j.aaltonen@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Fri, 14 Jan 2011 09:33:26 +0000 (10:33 +0100)]
ALSA: hda - More coverage for odd-number channels elimination for HDMI
The commit ad09fc9d2156f3d37537b34418a6b79309013d33 didn't cover the
case for Intel and Nvidia HDMIs, where hdmi_pcm_open() is called.
Put the hw_constraint there, too.
Takashi Iwai [Fri, 14 Jan 2011 09:30:46 +0000 (10:30 +0100)]
ALSA: hda - Store PCM parameters properly in HDMI open callback
In hdmi_pcm_open(), the evaluated PCM hw parameters are stored in
hinfo, but these aren't properly set back to the current runtime
record since these have been set beforehand in azx_pcm_open().
This patch fixes the behavior.
Takashi Iwai [Fri, 14 Jan 2011 08:42:27 +0000 (09:42 +0100)]
ALSA: hda - Suppress the odd number of channels for HDMI
It looks like that HDMI codecs don't support the odd number of channels
although HD-audio spec doesn't have the restriction. Add the
hw_constraint to limit to only the even number of channels.
Takashi Iwai [Thu, 13 Jan 2011 14:41:45 +0000 (15:41 +0100)]
ALSA: hda - Add fixup-call in init callback
In some cases, the fix-up is required in the init callback to be called
both at the first initialization and at the resume. The new action type
ALC_FIXUP_ACT_INIT is used for this case.
Takashi Iwai [Thu, 13 Jan 2011 13:22:32 +0000 (14:22 +0100)]
ALSA: hda - Reorganize fixup structure for Realtek
Instead of keeping various data types in a single record, put the
type field and keep a single value in each entry, but allows chaining
multiple fixup entries. This allows more flexible data management
(see ALC275_FIXUP_SONY_HWEQ for example).
Takashi Iwai [Tue, 11 Jan 2011 17:11:04 +0000 (18:11 +0100)]
ALSA: hda - Add static_hdmi_pcm option to HDMI codec parser
The dynamic PCM restriction based on ELD information may lead to the
problem in some cases, e.g. when the receiver is turned off. Then it
may send a TV HDMI default such as channels = 2. Since it's still
plugged, the driver doesn't know whether it's the right configuration
for future use. Now, when an app opens the device at this moment,
then turn on the receiver, the app still sends channels=2.
The right solution is to implement some kind of notification and
automatic re-open mechanism. But, this is a goal far ahead.
This patch provides a workaround for such a case by providing a new
module option static_hdmi_pcm for snd-hda-codec-hdmi module. When
this is set to true, the driver doesn't change PCM parameters per
ELD information. For users who need the static configuration like
the scenario above, set this to true.
The parameter can be changed dynamically via sysfs, too.
The DAPM code has been removed from the driver, but the
dapm struct remained.
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>
Peter Ujfalusi [Mon, 10 Jan 2011 13:39:49 +0000 (15:39 +0200)]
ASoC: tlv320dac33: Add DAPM selection for LOM invert
The L/R LOM line can be invertined side of the
corresponding DAC, or inverted from the corresponding
LOP.
Add control for user space to select the source of the
LOM inversion.
When only the analog bypass is enabled, and the LOM
is inverted from DAC output, we need to power the
corresponding DAC.
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>
David Lambert [Thu, 6 Jan 2011 14:00:37 +0000 (08:00 -0600)]
ASoC: DMIC codec: Adding a generic DMIC codec
This codec is to be used by the DMIC driver to
control the DMIC codec. This driver will be used on future
implementations of the DMIC driver to support codec specific
features.
At this time, the codec driver just registers the codec DAI.
Signed-off-by: David Lambert <dlambert@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
ASoC: soc-cache: Fix invalid memory access during snd_soc_lzo_cache_sync()
The size of the lzo syncing bitmap was incorrectly set to the size
of the cache times the word size, however, the correct size is the
size of the cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit 6d803ba736abb5e122dede70a4720e4843dd6df4 "ARM: 6483/1: arm & sh:
factorised duplicated clkdev.c" broke compilation of migor audio. Use the
correct header to fix the problem.
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>
Timur Tabi [Mon, 10 Jan 2011 19:28:32 +0000 (13:28 -0600)]
ASoC: cs4270: use the built-in register cache support
Update the CS4270 driver to use ASoC's internal codec register cache feature.
This change allows ASoC to perform the low-level I2C operations necessary to
read the register cache. Support is also added for initializing the register
cache with an array of known power-on default values.
The CS4270 driver was handling the register cache itself, but somwhere along
the conversion to multi-compaonent, this feature broke.
Signed-off-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Replace the get_i2s_mclk callback with tables of MCLK values. This
simplifies the MCLK-handling code in both the framework and the model-
specific drivers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove the DAC Oversampling mixer control because this setting does not
make much sense.
For cards with the H6 daughterboard, 128x oversampling was disabled
anyway because these high MCLK frequency would not be compatible with
the connector cable.
For cards without the H6 daughterboard, 128x gives a slightly higher
output quality; there is no reason to reduce it to 64x except for saving
power, but then these cards have not been designed to be power efficient
anyway (the D2's blinkenlights cannot be disabled).
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 10 Jan 2011 15:07:11 +0000 (16:07 +0100)]
ALSA: virtuoso: configure correct master clock frequency on the CS2000
The clock output of the CS2000, which is used as master clock for the
DACs, was using half the actual master clock frequency for some reason.
Using the theoretically correct frequency seems also to work in practice.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 10 Jan 2011 15:05:38 +0000 (16:05 +0100)]
ALSA: virtuoso: remove non-working controls on Essence ST Deluxe
On the Xonar Essence ST Deluxe, remove all mixer controls that would
require I2C communication with the third DAC, which does not work
because of an addressing conflict with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 10 Jan 2011 15:03:17 +0000 (16:03 +0100)]
ALSA: virtuoso: change PCM1796 format to I2S
Change the PCM format used for the PCM1796 from left-justified to I2S to
ensure that the correct format is used even for the Essence ST Deluxe's
center/LFE DAC, where I2C does not work because of an address conflict
with the CS2000 chip.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 10 Jan 2011 15:02:32 +0000 (16:02 +0100)]
ALSA: virtuoso: wait for PCM1796 clock to become stable
The PCM1796 needs the master clock for I2C communication to work, so
add delays after clock changes to ensure that the clock is stable when
we try to write the DACs' registers.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clemens Ladisch [Mon, 10 Jan 2011 14:59:38 +0000 (15:59 +0100)]
ALSA: oxygen: allow different number of PCM and mixer channels
For cards like the Xonar HDAV1.3, differentiate between the number of
PCM channels that can be played and the number of channels whose volume
can be adjusted.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai [Mon, 10 Jan 2011 14:45:23 +0000 (15:45 +0100)]
ALSA: hda - Add support for multiple headphone/speaker controls for Realtek
So far, Realtek auto-parser assumed that the multiple pins are only for
line-outs, and assigned the channel names like Front, Surround, etc for
the multiple outputs. But, there are devices that have multiple
headphones, and these can be better controlled with the corresponding
control-name like "Headphone" with indicies.
Takashi Iwai [Mon, 10 Jan 2011 13:47:35 +0000 (14:47 +0100)]
ALSA: hda - Fix multi-headphone handling for Realtek codecs
When multiple headphone pins are defined without line-out pins, the
driver takes them as primary outputs. But it forgot to set line_out_type
to HP by assuming there is some rest of HP pins. This results in some
mis-handling of these pins for Realtek codec parser. It takes as if
these are pure line-out jacks.
Jesper Juhl [Thu, 6 Jan 2011 21:19:47 +0000 (22:19 +0100)]
ALSA: Don't leak in sound/core/oss/pcm_oss.c::snd_pcm_hw_param_near()
snd_pcm_hw_param_near() will leak the memory allocated to 'save' if the
call to snd_pcm_hw_param_max() returns less than zero.
This patch makes sure we never leak.
Reported-and-tested-by: Fernando Lemos <fernandotcl@gmail.com> Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Add Lenovo vendor quirk for Conexant 205xx
BugLink: http://bugs.launchpad.net/bugs/689036
Many new Lenovos need the ideapad quirk. Also, since the
auto parser for this chip is far from optimal, the regression
risk is low (although not zero).
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Fix volume control indices for Mics (Realtek)
If more than one mic is present with different locations,
e g "Front Mic" and "Rear Mic", they can use the same index (0),
since their names are different.
Previous behavior was to have "Front Mic" as index 1, causing it
to be ignored by e g PulseAudio.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Rename "Mic Boost" to "Mic Boost Volume"
BugLink: http://bugs.launchpad.net/bugs/697240
If the "Volume" suffix is not given, alsa-lib gets confused and
loses the dB information at the simple element level.
Boosts generally affects both playback and capture, as they are
applied early in the chain. Hence no "Playback" or "Capture" in
the suffix.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/696493
According to datasheet (and real-world testing), IDT 92HD88B can
have internal mics at NID 0x11 and 0x20, so enable them accordingly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Harsha Priya [Wed, 5 Jan 2011 06:04:51 +0000 (11:34 +0530)]
ASoC: Fix the device references to codec and platform drivers
The soc-core takes the platform and codec driver reference during probe. Few of
these references are not released during remove. This cause the platform and
codec driver module unload to fail.
This patch fixes by the taking only one reference to platform and codec module
during probe and releases them correctly during remove. This allows load/unload
properly
Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Jarkko Nikula [Wed, 5 Jan 2011 10:05:44 +0000 (12:05 +0200)]
ASoC: Remove needless inclusion of tlv320aic3x.h from machine drivers
After multi-component conversion these machine drivers don't actually need
anything from sound/soc/codecs/tlv320aic3x.h so don't include it.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mark Brown [Sun, 2 Jan 2011 14:07:46 +0000 (14:07 +0000)]
ASoC: Change Samsung Kconfig from ASOC_ to SND_SOC_
The rest of ASoC is using SND_SOC_ as the prefix for all the Kconfig
symbols so do so for the new Samsung drivers too, rather than using
ASOC_ as they currently are.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Seungwhan Youn <sw.youn@samsung.com> Acked-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* git://git.kernel.org/pub/scm/linux/kernel/git/davem/net-2.6:
ipv4/route.c: respect prefsrc for local routes
bridge: stp: ensure mac header is set
bridge: fix br_multicast_ipv6_rcv for paged skbs
atl1: fix oops when changing tx/rx ring params
drivers/atm/atmtcp.c: add missing atm_dev_put
starfire: Fix dma_addr_t size test for MIPS
tg3: fix return value check in tg3_read_vpd()
Broadcom CNIC core network driver: fix mem leak on allocation failures in cnic_alloc_uio_rings()
ISDN, Gigaset: Fix memory leak in do_disconnect_req()
CAN: Use inode instead of kernel address for /proc file
skfp: testing the wrong variable in skfp_driver_init()
ppp: allow disabling multilink protocol ID compression
ehea: Avoid changing vlan flags
ueagle-atm: fix PHY signal initialization race
Joel Sing [Mon, 3 Jan 2011 20:24:20 +0000 (20:24 +0000)]
ipv4/route.c: respect prefsrc for local routes
The preferred source address is currently ignored for local routes,
which results in all local connections having a src address that is the
same as the local dst address. Fix this by respecting the preferred source
address when it is provided for local routes.
This bug can be demonstrated as follows:
# ifconfig dummy0 192.168.0.1
# ip route show table local | grep local.*dummy0
local 192.168.0.1 dev dummy0 proto kernel scope host src 192.168.0.1
# ip route change table local local 192.168.0.1 dev dummy0 \
proto kernel scope host src 127.0.0.1
# ip route show table local | grep local.*dummy0
local 192.168.0.1 dev dummy0 proto kernel scope host src 127.0.0.1
We now establish a local connection and verify the source IP
address selection:
Thus without the patch, with the boot parameters 'tcb selinux=0', adding
the above 'dont_measure obj_type=var_log_t' rule to the default IMA TCB
measurement policy, would result in nothing being measured. The patch
prevents the default TCB policy from being replaced.
Signed-off-by: Mimi Zohar <zohar@us.ibm.com> Cc: James Morris <jmorris@namei.org> Acked-by: Serge Hallyn <serge.hallyn@canonical.com> Cc: David Safford <safford@watson.ibm.com> Cc: <stable@kernel.org> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
This seems fine itself, but br_send_bpdu() invokes ebtables LOCAL_OUT.
We oops in ebt_basic_match() because it assumes eth_hdr(skb) returns
a meaningful result.
Cc: acme@ghostprotocols.net
References: https://bugzilla.kernel.org/show_bug.cgi?id=24532 Signed-off-by: Florian Westphal <fw@strlen.de> Signed-off-by: David S. Miller <davem@davemloft.net>