* 'master' of git://git.alsa-project.org/alsa-kernel:
[ALSA] intel8x0: add one retry to the ac97_clock measurement routine
[ALSA] intel8x0: fix wrong conditions in ac97_clock measure routine
[ALSA] intel8x0: do not use zero value from PICB register
[ALSA] intel8x0: an attempt to make ac97_clock measurement more reliable
[ALSA] pcm-midlevel: Add more strict buffer position checks based on jiffies
[ALSA] hda_intel: fix unexpected ring buffer positions
Added the models for quirk bitmask 1734:110x and 1734:113x of
Fujitsu laptops.
This will fix the model detection for Amilo Xa3540.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that on some hardware platforms, the first measurement is wrong.
This patch adds second measurement to this case.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Currently there are two possible platform datas for the PXA AC97 driver:
one supported by the generic AC97 driver only which provides callbacks
to allow board-specific configuration at stream startup and teardown,
and another for pxa2xx-ac97-lib which allows configuration of the reset
GPIO for PXA2xx CPUs.
Obviously this won't actually work when using the generic AC97 driver
since the drivers will attempt to parse the platform data in both
formats. Fix this by merging the two structures.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Robert Jarzmik <robert.jarzmik@free.fr>
Cc: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Don't call snd_jack_report at release of sigmatel and conexnat codecs
which results in Oops at unloading the module.
The Oops is triggered by the power-up sequence during the free due to
the pincfg restoration. Since the power-up sequence is involved with
the unsol handling, the jack reporting may be issued during that.
The Oops occurs with this jack reporting because the jack instances
have been already released but the codec doesn't do the proper
book-keeping.
This patch adds the book-keeping of jack instances to avoid the access
to bogus pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added private_data and private_free fields to struct snd_jack so that
the caller can assign the data. It'll be helpful for avoiding the
double-free of the jack instance.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This is the second go through of the old DMA_nBIT_MASK macro,and there're not
so many of them left,so I put them into one patch.I hope this is the last round.
After this the definition of the old DMA_nBIT_MASK macro could be removed.
Signed-off-by: Yang Hongyang <yanghy@cn.fujitsu.com>
Cc: Russell King <rmk@arm.linux.org.uk>
Cc: Tony Lindgren <tony@atomide.com>
Cc: "David S. Miller" <davem@davemloft.net>
Cc: James Bottomley <James.Bottomley@HansenPartnership.com>
Cc: Greg KH <greg@kroah.com>
Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
It seems that the zero value from the PICB (position in current buffer)
register is not reliable. Use jiffies to correct returned value
from the ring buffer pointer callback.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- use monotonic posix clock to measure time
- try to avoid reading zero from PICB (position in current buffer) register
- show also measured samples
- when clock is near 41000 or 44100, use exactly these values
(they appears to be reference clocks for hardware manufacturers)
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
The WM8960 is a low power, high quality stereo codec designed for
portable digital audio applications.
Stereo class D speaker drivers provide 1W per channel into 8W loads.
Guaranteed low leakage, excellent PSRR and pop/click suppression
mechanisms enable direct battery connection for the speaker supply.
The device also integrates a complete microphone interface and a stereo
headphone driver. External component requirements are drastically
reduced as no separate microphone, speaker or headphone amplifiers are
required. Advanced on-chip digital signal processing performs automatic
level control for the microphone or line input.
Stereo 24-bit sigma-delta ADCs and DACs are used with low power
over-sampling digital interpolation and decimation filters and a
flexible digital audio interface.
The master clock can be input directly or generated internally by an
onboard PLL, supporting most commonly-used clocking schemes.
This driver was originally written by Liam Girdwood, with substantial
subsequent additions and updates for feature completeness and changes in
the ASoC framework from me.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)
SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).
This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This ensures that we sync with the DAPM powerdown sequencing properly
and don't need to bounce the power on the voice DAC so often.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is simple code motion, intended to support future refactoring of
the DAPM algorithms and (more immediately) the additon of events for
DACs and ADCs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some drivers like Intel8x0 or Intel HDA are broken for some hardware variants.
This patch adds more strict buffer position checks based on jiffies when
internal hw_ptr is updated. Enable xrun_debug to see mangling of wrong
positions.
As a side effect, the hw_ptr interrupt update routine might do slightly better
job when many interrupts are lost.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
I found two issues with ICH7-M (it should be related to other HDA chipsets
as well):
- the ring buffer position is not reset when stream restarts (after xrun) -
solved by moving azx_stream_reset() call from open() to prepare() callback
and reset posbuf to zero (it might be filled with hw later than position()
callback is called)
- irq_ignore flag should be set also when ring buffer memory area is not
changed in prepare() callback - this patch replaces irq_ignore with
more universal check based on jiffies clock
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Due to the process and communications issues with the 2.6.30 S3C
platform merges none of the underlying arch/arm code for S3C64xx audio
support made it into mainline, rendering the drivers useless. Disable
them in Kconfig to avoid user confusion - users patching in the required
support can always reenable this too.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Eric Miao <eric.miao@marvell.com>
Cc: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8988 is a low power, high quality stereo CODEC designed for
portable digital audio applications.
The device integrates complete interfaces to 2 stereo headphone or line
out ports. External component requirements are drastically reduced as no
separate headphone amplifiers are required. Advanced on-chip digital
signal processing performs graphic equaliser, 3-D sound enhancement and
automatic level control for the microphone or line input.
The WM8988 can operate as a master or a slave, with various master clock
frequencies including 12 or 24MHz for USB devices, or standard 256fs
rates like 12.288MHz and 24.576MHz. Different audio sample rates such as
96kHz, 48kHz, 44.1kHz are generated directly from the master clock
without the need for an external PLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.
A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (36 commits)
ALSA: hda - Add VREF powerdown sequence for another board
ALSA: oss - volume control for CSWITCH and CROUTE
ALSA: hda - add missing comma in ad1884_slave_vols
sound: usb-audio: allow period sizes less than 1 ms
sound: usb-audio: save data packet interval in audioformat structure
sound: usb-audio: remove check_hw_params_convention()
sound: usb-audio: show sample format width in proc file
ASoC: fsl_dma: Pass the proper device for dma mapping routines
ASoC: Fix null dereference in ak4535_remove()
ALSA: hda - enable SPDIF output for Intel DX58SO board
ALSA: snd-atmel-abdac: increase periods_min to 6 instead of 4
ALSA: snd-atmel-abdac: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: replace bus_id with dev_name()
ALSA: snd-atmel-ac97c: cleanup registers when removing driver
ALSA: snd-atmel-ac97c: do a proper reset of the external codec
ALSA: snd-atmel-ac97c: enable interrupts to catch events for error reporting
ALSA: snd-atmel-ac97c: set correct size for buffer hardware parameter
ALSA: snd-atmel-ac97c: do not overwrite OCA and ICA when assigning channels
ALSA: snd-atmel-ac97c: remove dead break statements after return in switch case
ALSA: snd-atmel-ac97c: cleanup register definitions
...
Replace all DMA_24BIT_MASK macro with DMA_BIT_MASK(24)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_28BIT_MASK macro with DMA_BIT_MASK(28)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_30BIT_MASK macro with DMA_BIT_MASK(30)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_31BIT_MASK macro with DMA_BIT_MASK(31)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)
Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
Fix for compillation error introduced by the constrain patch.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add powerdown sequence for VREF using a shared jack when the headphone
is present and the microphone isn't on.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added an else part to check
SNDRV_MIXER_OSS_PRESENT_CVOLUME for MIC (slot 7)
in commit 36c7b833e5
Similarly, checks and volume control is required for
SNDRV_MIXER_OSS_PRESENT_CSWITCH and SNDRV_MIXER_OSS_PRESENT_CROUTE
as well.
Signed-off-by: Deepika Makhija <deepika.makhija@einfochips.com>
Signed-off-by: Viral Mehta <viral.mehta@einfochips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To enable periods shorter than 1 ms, we have to make sure that short
periods are only available for alternate settings that have a small
enough data packet interval. Furthermore, the code that aligns URBs to
USB frames is now superfluous.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The data packet interval needs to be available in the audioformat
structure, together with the other audio format parameters, so that it
can be used to influence ALSA hardware parameters.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This removes the check_hw_params_convention() function because
1) it is not necessary, as the hw_rule_* functions also work correctly
(i.e., as no-ops) when the device supports all combinations of the
audio format parameters; and
2) it would become too complex when adding a fourth altsetting-dependent
hardware parameter, as this would require another three loops to
check dependecies with rate/channels/format.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When listing the device's sample formats in the stream? proc file, the
sample format number itself is rather obscure, so we better show the
format width, too.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The driver should pass a device that specifies internal DMA ops, but
substream->pcm is just a logical device, and thus doesn't have arch-
specific dma callbacks, therefore following bug appears:
Freescale Synchronous Serial Interface (SSI) ASoC Driver
------------[ cut here ]------------
kernel BUG at arch/powerpc/include/asm/dma-mapping.h:237!
Oops: Exception in kernel mode, sig: 5 [#1]
...
NIP [c02259c4] snd_malloc_dev_pages+0x58/0xac
LR [c0225c74] snd_dma_alloc_pages+0xf8/0x108
Call Trace:
[df02bde0] [df02be2c] 0xdf02be2c (unreliable)
[df02bdf0] [c0225c74] snd_dma_alloc_pages+0xf8/0x108
[df02be10] [c023a100] fsl_dma_new+0x68/0x124
[df02be20] [c02342ac] soc_new_pcm+0x1bc/0x234
[df02bea0] [c02343dc] snd_soc_new_pcms+0xb8/0x148
[df02bed0] [c023824c] cs4270_probe+0x34/0x124
[df02bef0] [c0232fe8] snd_soc_instantiate_card+0x1a4/0x2f4
[df02bf20] [c0233164] snd_soc_instantiate_cards+0x2c/0x68
[df02bf30] [c0234704] snd_soc_register_platform+0x60/0x80
[df02bf50] [c03d5664] fsl_soc_platform_init+0x18/0x28
...
This patch fixes the issue by using card's device instead.
Signed-off-by: Anton Vorontsov <avorontsov@ru.mvista.com>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Always use request_firmware() for loading yss225_registers image.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: David Woodhouse <David.Woodhouse@intel.com>
According to the data sheet data is clocked out on the falling edge
and latched on the rising edge of the bit clock. While the left sample
is transmitted the word clock line is low.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ak4535_remove() from sound/soc/codecs/ak4535.c calls
i2c_unregister_device() with a possibly null pointer.
This bug was found by smatch (http://repo.or.cz/w/smatch.git/).
Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both.
DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add
0x10 as slave output to the existing 0x06.
Reported-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Tested-by: Jeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch increases periods_min to 6 from 4, this will remove any
hickups where the buffer is not filled fast enough from user space.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch replaces the references to bus_id to the new dev_name() API.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch replaces the references to bus_id to the new dev_name() API.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch will set the channel A and control channel mode register to
zero before disabling the AC97C peripheral.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch will enable the AC97C before resetting the external codec,
leaving the AC97C disabled will result in floating I/O lines that can
affect the reset procedure.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch will enable interrupts from AC97C and report about error
conditions that occurs.
On channel A both overrun and underrun will be enabled depending if
playback and/or capture are enabled. On the control channel the overrun
interrupt is enabled.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch will set a proper maximum bytes for the buffer, which is:
channels * bytes per sample * maximum periods * maximum bytes per period.
It also sets the minimum periods to 6, a value chosen from testing, with
a minimum of 6 periods the system has good time to fill in new audio
data without skipping.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch will take care not to overwrite OCA and ICA registers when
assigning input and output channels. It will also make sure the
registers are at a known state when enabling a channel and clean up
properly in case of an error.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch will remove traces of channel B registers, since they are not
used by the AC97C driver. Channel B might be used for other purposes.
The driver also adds channel status bits TXEMPTY and OVRUN and a
AC97C_CH_MASK macro to ease clearing a channel settings.
Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added device id in struct for codec 92HD81B1C (0x111d76d5).
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A pointer to snd_pmac_probe is passed to the core via
platform_driver_register and so the function must not disappear when the
.init sections are discarded. Otherwise (when having HOTPLUG=y)
unbinding and binding a device to the driver via sysfs will result in an
oops as does a device being registered late.
An alternative to this patch is using platform_driver_probe instead of
platform_driver_register plus removing the pointer to the probe function
from the struct platform_driver.
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Johannes Berg <johannes@sipsolutions.net>
Cc: Rene Herman <rene.herman@keyaccess.nl>
Cc: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add ZV port control switch.
This patch is done after solution
given in the ALSA bug #2872 report.
The patch resolves the issue.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds machine specific code for the audio part of the Stretch
s6105 IP camera reference design.
The device uses the tlv320aic31(01) codec to generate the clock for
both I2S ports of the soc. While the master clock is generated by a
configurable PLL chip, the code assumes the factory default settings.
An additional kcontrol has been added to handle the special routing of
the board, connecting both HPLCOM and HPROUT to the same pin of the audio
jack. One of these should always be switched off.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds a driver for the I2S interface found on Stretch s6000
family processors.
Signed-off-by: Daniel Glöckner <dg@emlix.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jikos/trivial: (28 commits)
trivial: Update my email address
trivial: NULL noise: drivers/mtd/tests/mtd_*test.c
trivial: NULL noise: drivers/media/dvb/frontends/drx397xD_fw.h
trivial: Fix misspelling of "Celsius".
trivial: remove unused variable 'path' in alloc_file()
trivial: fix a pdlfush -> pdflush typo in comment
trivial: jbd header comment typo fix for JBD_PARANOID_IOFAIL
trivial: wusb: Storage class should be before const qualifier
trivial: drivers/char/bsr.c: Storage class should be before const qualifier
trivial: h8300: Storage class should be before const qualifier
trivial: fix where cgroup documentation is not correctly referred to
trivial: Give the right path in Documentation example
trivial: MTD: remove EOL from MODULE_DESCRIPTION
trivial: Fix typo in bio_split()'s documentation
trivial: PWM: fix of #endif comment
trivial: fix typos/grammar errors in Kconfig texts
trivial: Fix misspelling of firmware
trivial: cgroups: documentation typo and spelling corrections
trivial: Update contact info for Jochen Hein
trivial: fix typo "resgister" -> "register"
...
Adds the needed code to be able to use 96KHz playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Without this the WM9705 driver fails badly when resuming.
Tested-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that any AC97 devices that bind to the CODEC are below the
ASoC device in the device tree so the suspend and resume code can
figure out what order to handle them in.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
AC97 devices may have other drivers hanging off them directly so need to
have resumed when the resume function returns meaning that we can't defer
the resume - complete it immediately for them. Non-AC97 devices should
not have other drivers hanging directly off the ASoC devices.
We only really need the deferral for non-AC97 devices - it's there since
some I2C buses are very slow and non-AC97 codecs often have large numbers
of registers to restore and require delays to bring the codec up cleanly
leading to a substantial impact on overall resume time.
Reported-by: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McBSP2 in OMAP3 has 1 ksample (1k x 32 bit) internal FIFO. During
initial playback startup, this FIFO is keeping the DMA request active
until the FIFO is full.
So now if ALSA buffer size is smaller, DMA is looping around it while
filling up the HW FIFO, generating burst of interrupts as well and SW
doesn't have any change to fill enough data.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In case of duplex mode (capture and playback at the same time), the second
stream has to have the same parameters (rate, sample size) as the already
running stream.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
TWL4030 supports 96KHz sample playback, but only playback.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Optimize the display of SSI statistics in the Freescale MPC8610 sound driver
to display the status count only of the interrupts that were actually enabled.
Previously, it would display the counts of all SISR status bits, even those
that were not enabled.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
the variable gsr_bit is set in isr. It is however set to 0 and interrupts are
disabled prior to reset. Hence it doesn't make a lot of sense to show the
content of gsr_bit in case of a reset timeout.
Signed-off-by: Luotao Fu <l.fu@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Remove the delay from the trigger function in the Freescale MPC8610 sound
driver when capture is started. This delay was used to ensure that the DMA
controller was active when ALSA call the .pointer function to request a
DMA transfer status. A better approach is for the .pointer function to detect
that DMA has not started, and return zero instead. This change eliminates
the need for the delay.
Also add some related code to check for a DMA programming error, and report
XRUN if it occurs.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
HTC Magician has a Philips UDA1380 codec connected via
SSP1 (playback) and I2S (capture).
There is a flip-flop between the SSP frame clock output
and the codec's word select input pin. To make the codec
see proper I2S input, the SSP has to send two frames per
sample.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now magician and similar boards can use network mode with only one
active slot to explicitly set 16 bit frame width, even for S16_LE
stereo sound.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Setting ->owner as done currently (pde->owner = THIS_MODULE) is racy
as correctly noted at bug #12454. Someone can lookup entry with NULL
->owner, thus not pinning enything, and release it later resulting
in module refcount underflow.
We can keep ->owner and supply it at registration time like ->proc_fops
and ->data.
But this leaves ->owner as easy-manipulative field (just one C assignment)
and somebody will forget to unpin previous/pin current module when
switching ->owner. ->proc_fops is declared as "const" which should give
some thoughts.
->read_proc/->write_proc were just fixed to not require ->owner for
protection.
rmmod'ed directories will be empty and return "." and ".." -- no harm.
And directories with tricky enough readdir and lookup shouldn't be modular.
We definitely don't want such modular code.
Removing ->owner will also make PDE smaller.
So, let's nuke it.
Kudos to Jeff Layton for reminding about this, let's say, oversight.
http://bugzilla.kernel.org/show_bug.cgi?id=12454
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (422 commits)
[ARM] 5435/1: fix compile warning in sanity_check_meminfo()
[ARM] 5434/1: ARM: OMAP: Fix mailbox compile for 24xx
[ARM] pxa: fix the bad assumption that PCMCIA sockets always start with 0
[ARM] pxa: fix Colibri PXA300 and PXA320 LCD backlight pins
imxfb: Fix TFT mode
i.MX21/27: remove ifdef CONFIG_FB_IMX
imxfb: add clock support
mxc: add arch_reset() function
clkdev: add possibility to get a clock based on the device name
i.MX1: remove fb support from mach-imx
[ARM] pxa: build arch/arm/plat-pxa/mfp.c only when PXA3xx or ARCH_MMP defined
Gemini: Add support for Teltonika RUT100
Gemini: gpiolib based GPIO support v2
MAINTAINERS: add myself as Gemini architecture maintainer
ARM: Add Gemini architecture v3
[ARM] OMAP: Fix compile for omap2_init_common_hw()
MAINTAINERS: Add myself as Faraday ARM core variant maintainer
ARM: Add support for FA526 v2
[ARM] acorn,ebsa110,footbridge,integrator,sa1100: Convert asm/io.h to linux/io.h
[ARM] collie: fix two minor formatting nits
...
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/geert/linux-m68k:
m68k: irq_node.handler() should return irqreturn_t
m68k: section mismatch fixes: Atari SCSI
m68k: section mismatch fixes: DMAsound for Atari
MAINTAINERS: Replace dead link to m68k CVS repository by link to new git repository
m68k: mac - Add SWIM floppy support
m68k: mac - Add a new entry in mac_model to identify the floppy controller type.
m68k: Add install target
* 'bkl-removal' of git://git.lwn.net/linux-2.6:
Rationalize fasync return values
Move FASYNC bit handling to f_op->fasync()
Use f_lock to protect f_flags
Rename struct file->f_ep_lock
* git://git.kernel.org/pub/scm/linux/kernel/git/lethal/sh-2.6: (96 commits)
sh: add support for SMSC Polaris platform
sh: fix the HD64461 level-triggered interrupts handling
sh: sh-rtc wakeup support
sh: sh-rtc invalid time rework
sh: sh-rtc carry interrupt rework
sh: disallow kexec virtual entry
sh: kexec jump: fix for ftrace.
sh: kexec: Drop SR.BL bit toggling.
sh: add kexec jump support
sh: rework kexec segment code
sh: simplify kexec vbr code
sh: Flush only the needed range when unmapping a VMA.
sh: Update debugfs ASID dumping for 16-bit ASID support.
sh: tlb-pteaex: Kill off legacy PTEA updates.
sh: Support for extended ASIDs on PTEAEX-capable SH-X3 cores.
sh: sh7763rdp: Change IRQ number for sh_eth of sh7763rdp
sh: espt-giga board support
sh: dma: Make G2 DMA configurable.
sh: dma: Make PVR2 DMA configurable.
sh: Move IRQ multi definition of DMAC to defconfig
...
Add a quirk model=acer-aspire for Acer Ferrari 5000 with ALC883 codec.
Note that model=auto doesn't work for this laptop because of broken BIOS
(that doesn't set the subsystem id properly).
Tested-by: Russ Dill <russ.dill@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Replace with the standard function calls to use caches for reading
the widget caps and pin caps.
hda_proc.c is still using the direct verbs to get raw values as
much as possible.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In patch_realtek.c, don't create empty or single-item "Input Source"
control elements that are simply superfluous.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The check for the amp-output must be done for widget-caps rather than
pin-caps as implemented in the recent change... Simply a thinko.
Also, add the similar checks to all places that put output-amp mutes
in the initialization.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_hda_query_pin_caps() to read and cache pin-cap values
to avoid too frequently issuing the same verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't set amp-out values to pins without PINCAP_OUT capability,
which are usually assigned for digital mics on ALC663/ALC272.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch does two things:
Output Intel HDA Function Id in /proc/asound/cardX/codec#X
Align Vendor/Subsystem/Revision Ids to 8 characters, front-padded with zeros
Before:
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x100100
After:
Function Id: 0x1
Vendor Id: 0x11d41884
Subsystem Id: 0x103c281a
Revision Id: 0x0100100
As report on the Kernel Bugzilla #12888
Signed-off-by: Pascal de Bruijn <pascal@unilogicnetworks.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make the boundary checks a bit safer.
These caese are rare or theoretically won't happen, but nothing
bad to keep the checks safer...
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the detection of digital-mic inputs on ALC663 / ALC272 codecs
in the auto-detection mode. The automatic mic switch via plugging
isn't implemented yet, though.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Headset was declared previously as a Headphone widget connecting
HSMIC and HSOL/HSOR pins of TWL4030 codec in SDP430 machine driver.
The capture path becomes invalid as the Headphone widget is not a
valid input endpoint.
Instead of that, the Headset is declared as separate Microphone
and Headphone widgets. Current patch modifies audio map:
- Headset Mic: HSMIC with bias
- Headset Stereophone: HSOL, HSOR
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add functions "Headset" and "Mic" to the control "Jack Function" for
activating and de-activating codec input pin LINE1L which is connected to
the mic pin of 4-pole Nokia AV connecter.
Note there is no mic bias voltage management here since bias is coming from
Nokia ASIC and driver for it is not in mainline.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
So far, the prepare callback is called multiple times, BDL entries
are reset and re-programmed at each time.
This patch adds the check to avoid the reset of BDL entries when the
same parameters are used.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the hw_ptr_interrupt reaches the boundary, it must check whether
the hw_base was already lapped and corret the delta value appropriately.
Also, rebasing the hw_ptr needs a correction because buffer_size isn't
always aligned to period_size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Always reset the invalind hw_ptr position returned by the pointer
callback. The behavior should be consitent independently from the
debug option.
Also, add the printk_ratelimit() check to avoid flooding debug
prints.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an AVDD supply as well, normally one or more of the other
upplies would be tied to it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The active discharge does not bring sufficient benefit to justify the
lengthy times involved so don't do that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Drop inbound packets that are smaller than expected. This has been
observed at the very beginning of the streaming transaction.
And when the hardware is in panic mode (which can only very rarely
happen in case of massive EMI chaos), mute the input channels.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Tested-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Limit the number of printed warnings to one in case of streaming errors.
printk() happens to be expensive, especially in code called as often as
here.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In snd_free_sgbuf_pags(), vunmap() is called after releasing the SG
pages, and it causes errors on Xen as Xen manages the pages
differently. Although no significant errors have been reported on
the actual hardware, this order should be fixed other way round,
first vunmap() then free pages.
Cc: Jan Beulich <jbeulich@novell.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
s/mutex_lock/mutex_unlock/ on 2 fail paths in snd_pcm_oss_proc_write.
Probably a typo, lock should be unlocked when leaving the function.
Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not
set for MIC (slot 7).
So, the same should not be checked when an application tries to do gain
control for audio recording devices.
Just check slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently.
Verified with a simple application which opens /dev/dsp for recording and
/dev/mixer for volume control.
Have tested two usb audio mic devices.
Signed-off-by: Viral Mehta <viral.mehta@einfochips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.
Also, clean a bit the code.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA. As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got
merged into mainline. Since there's no way to use it on any of
supported machines (iPaq h3100 or h3600), better drop it for now.
It can be reimplemented later using ASoC infrastructure (there's
already a driver for uda1341 codec in mainline, so only CPU and machine
parts need to be written).
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Cc: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 14fa43f53f ("ASoC: Only
register AC97 bus if it's not done already") added a condition for
calling of soc_ac97_dev_register() but not added for calling of
soc_ac97_dev_unregister(). This patch adds same condition for
soc_ac97_dev_unregister(). Without this fix, kernel crashes when
unloading an asoc driver.
Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the OOPS during a opl3sa2 card suspend
and resume if the driver is loaded but the card
is not found.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Most fasync implementations do something like:
return fasync_helper(...);
But fasync_helper() will return a positive value at times - a feature used
in at least one place. Thus, a number of other drivers do:
err = fasync_helper(...);
if (err < 0)
return err;
return 0;
In the interests of consistency and more concise code, it makes sense to
map positive return values onto zero where ->fasync() is called.
Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
Traditionally, changes to struct file->f_flags have been done under BKL
protection, or with no protection at all. This patch causes all f_flags
changes after file open/creation time to be done under protection of
f_lock. This allows the removal of some BKL usage and fixes a number of
longstanding (if microscopic) races.
Reviewed-by: Christoph Hellwig <hch@lst.de>
Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
CC sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.
This patch removes the line-out volume control from vmixer-equipped cards.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As the PXA27x series allow 2 gpios to reset the ac97 bus,
allow through platform data configuration the definition of
the correct gpio which will reset the AC97 bus.
This comes from a silicon defect on the PXA27x series, where
the gpio must be manually controlled in warm reset cases.
Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Two issues are fixed here:
- I2S transmits the left frame with the clock low but I don't seem to
get LRCLK out without SFRMDLY being set so invert SFRMP and set a
delay.
- I2S has a clock cycle prior to the first data byte in each channel
so we need to delay the data by one cycle.
Tested-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This switches the pxa ssp port usage from network mode to PSP mode.
Removed some comments and checks for configured TDM channels.
A special case is added to support configuration where BCLK = 64fs. We
need to do some black magic in this case which doesn't look nice but
there is unfortunately no other option than that.
Diagnosed-by: Tim Ruetz <tim@caiaq.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move headset jack registration to the codec/machine specific
initialization. Having the jack registration in machine init
causes that the jack device gets initialized but not registered
since the sound card is registered before the jack. Moving jack
registration to device initialization will register the jack
device along with all other devices associated to the card when
the card is registed. As a consequence of jack device registered
properly, the jack is detected as an input device.
Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.
Also, keep power-up during hwdep reconfiguration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.
Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.
Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line. Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application. This driver supports the
primary audio CODEC features, including:
- 1W speaker driver
- Fully differential headphone output
- Up to 4 differential microphone inputs
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Buildfix:
CC sound/soc/omap/osk5912.o
sound/soc/omap/osk5912.c: In function 'osk_soc_init':
sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount'
make[3]: *** [sound/soc/omap/osk5912.o] Error 1
There's no such (standard) clock interface.
Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all.
If there would be but the SSP port is in use already, bail out.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Clean up Conexant 5047 pareser code:
- Split mixer elements to separate arrays to reduce the duplicated
entires
- Fix mixer element names to the standard ones
- Remove unneeded cxt5047_hp2_unsol_event; the normal unsol_event
handler works fine.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the initial connections of output pins 0x13 and 0x1d for Conexant
5047 codec to point to the mixer amp properly.
Removed unneeded (doubly) verbs from arrays, also removed the unneeded
changing of widget 0x1c, which is now completely unused.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove superfluous verbs from cxt5047_toshiba_init_verbs[].
Also fix comments and minor coding style issues.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create "Capture Source" control dynamically for Conexant codecs.
If only one capture item is available, don't create such a control
since it's just useless.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of binding volumes, create a virtual master volume for Conexant
codecs. This allows separate HP and speaker volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Code rework, comments of mail tiwai@suse.de (2009-03-09) incorporated.
Code tested on HP HDX16 (not tested on HDX18 yet).
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added codec recognition of HP HDX platforms and added support of the
MUTE LED (orange/white). For this feature the CONFIG_SND_HDA_POWER_SAVE
is needed to use event handling for mute control.
Signed-off-by: Christoph Plattner <christoph.plattner@gmx.at>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This will break any boards that don't register the AC97 controller
device due to using ASoC.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use the standard linked list for snd_monitor_file management.
Also, move the list deletion of shutdown_list element into
snd_disconnect_release() (for simplification).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added snd_ctl_add_slave_uncached() function to add a slave element
with volatile controls. The values of normal slave elements are
supposed to be cachable, i.e. they are changed only via the put
callbacks. OTOH, when a slave element is volatile and its values may
be changed by other reason (e.g. hardware status change), the values
will get inconsistent.
The new function allows the slave elements with volatile changes.
When the slave is tied with this call, the native get callback is
issued at each time so that the values are always updated.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
pxa-regs.h and hardware.h are not intended for use directly in driver
code, remove those unnecessary references.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
1. Driver code where pxa_request_dma() is called will most likely
reference DMA registers as well, and it is really unnecessary
to include pxa-regs.h just for this. Move the definitions into
<mach/dma.h> and make relevant drivers include it instead of
<mach/pxa-regs.h>.
2. Introduce DMAC_REGS_VIRT as the virtual address base for these
DMA registers. This allows later processors to re-use the same
IP while registers may start at different I/O address.
Signed-off-by: Eric Miao <eric.miao@marvell.com>
The timer callbacks are called in the protected status by the lock
of the timer instance, so there is no need for an extra lock in the
PCM substream.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Clean up and improve snd_pcm_update_hw_ptr*() functions.
snd_pcm_update_hw_ptr() tries to detect the unexpected hwptr jumps
more strictly to avoid the position mess-up, which often results in
the bad quality I/O with pulseaudio.
The hw-ptr skip error messages are printed when xrun proc is set to
non-zero.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds a driver for the SPI connected AK4104 S/PDIF transmitter
device. Its features are fairly simple, but as there is need to set up
certain bits in the IEC958 information, this better goes into a real
driver.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Mark Brown <broonie@sirena.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On the HT-Omega Claro (halo) sound cards, the headphone amplifier must
be enabled explicitly by setting a GPIO bit.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Removes numbers from the list of features/limitations and makes it
reflect recent changes to the code.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Fix headphone-detect regression with multiple HP jacks
ALSA: hda - Fix typos in slave controls in patch_sigmatel.c
