Since the SND_SOC_DAPM_LINE can be input or output, additional check is
needed in order to determine if the widget is connected as input or
output.
When checking for connected outputs, if the widget is line, than check
if the sources list is not empty (line is connected as output)
For input endpoint check, when the widget is line, also check if the
sinks list is not empty (line is connected as input).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The remove callback has to be marked as __devexit, as the dynamic unbind
is possible.
Reported-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-parser for ALC662/663/272 codecs doesn't work properly when
a speaker is connected to mono NID 0x17, and doesn't handle the dynamic
DAC assignment properly.
This patch fixes the issues and also improves the assignment of DACs
so that HP and speakers can have independent volume controls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On Soundblaster X-FI Titenium with emu20k2 the SIDE and SURROUND mute
functions are swapped.
It was checked with 'speaker-test -c 8 -s 3' and (un)mute surround or
'speaker-test -c 8 -s 7' and (un)mute side. The volume seems not
to be affected and works as expected.
Signed-off-by: Sven Eckelmann <sven.eckelmann@gmx.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I can't see any reason for struct i2c_driver keywest_driver to not be
static.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/410933
This Sony VAIO model also needs External Amplifier unmuted for audible
playback, so make sure we set the inv_eapd quirk.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the auto-mic switching between an analog and a digital mic is
needed with IDT codecs, the current driver doesn't reset the connection
of the digital mux.
This patch fixes the behavior by checking both mux connections properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:
debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
/dapm_pop_time
/dapm/{widgets}
With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some of the Blackfin options don't directly follow the kconfig options
they depend on, so kconfig is unable to display the proper tree. So sort
the options such they expand/collapse properly.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Patch was tested on Toshiba NB200 and is found to enable sound.
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the missing argument of snd_soc_dai_set_pll() in neo1973_*.c,
which was forgotten in the commit 85488037bb.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The client->driver pointer can be NULL when i2c-device probing fails
in i2c_new_device(). This patch adds the NULL checks for client->driver
and return the error instead of blind assumption of driver availability.
Reported-by: Tim Shepard <shep@alum.mit.edu>
Cc: Jean Delvare <khali@linux-fr.org>
Cc: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/410933
This Sony VAIO model needs External Amplifier unmuted for audible
playback, so make sure we set the inv_eapd quirk.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a couple of typos and a missing header file inclusion to build wm8711.c
properly with CONFIG_SPI_MASTER.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Don't use a static for WM8974 PLL factors - we don't support more than
one device so it won't happen but no sense in leaving the race condition
hanging around. Also, pre_div is a single bit and it's a bit simpler if
we move the handling of the factor of 4 in the output into the
coefficient setup.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The DMA params for McASP with FIFO has been updated so that it works for
various FIFO levels. A member- 'fifo_level' has been added to the DMA
params data structure. The fifo_level can be adjusted by the tx[rx]_numevt
platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This
implementation has been tested for numevt values 1, 2, 4, 8.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Mia has an undocumented line-out control, and it has to be exposed.
Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit fdbc66266c, I mistakenly
replaced the capture mixer array for ALC268_ACER to nosrc version
although this should be kept to alt_mixer. Now fixed back.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reference: ALSA bug #0004614https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614
port-A (0x11) - front hp-out
port-D (0x12) - rear line out
port-E (0x1c) - front mic-in
port-F (0x16) - Internal speakers
digital-mic (0x17) - Internal mic
init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware
Signed-off-by: Miguel de Barros <miguel.de.barros@bluewin.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* mark struct vm_area_struct::vm_ops as const
* mark vm_ops in AGP code
But leave TTM code alone, something is fishy there with global vm_ops
being used.
Signed-off-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
When running in TDM mode there can be more than 2 channels used. Datasheet has
figures for upto 8 channels so increase max_channels on all SSP interfaces to
this figure.
Signed-off-by: Graeme Gregory <dp@xora.org.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
After puting a cd-audio inside my laptop there was no sound out here,
so I decided to install alsa-driver with debug level and setup a
model=test, it didn't help, but then I look at source code and added
this few lines, now cd-audio is working both when playback/recording.
[Additional minor fixes of mixer element/item names by tiwai]
Signed-off-by: Lukasz Marcinowski <nowymarluk@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
McASP write FIFO registers should be modified for playback and read FIFO
registers for capture. Check the PCM mode before manipulating the
FIFO registers. Currently, irrespective of playback/capture both the
FIFOs are enabled or disbaled. This resulted in errors in audio loopback
mode.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch removes references to cpu_dai->dma_data.
It makes struct davinci_pcm_dma_params part of
struct davinci_mcbsp_dev or struct davinci_audio_dev.
It removes the unused name variable from davinci_pcm_dma_params.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When both playback and capture stream were open
davinci_i2s_hw_params was setting parameters for
the wrong stream. The fix for davinci_i2s_hw_params
is sufficient, but it looks like a race still happens
in davici_pcm_open. This patch also makes the race smaller
but the next patch provides a better fix.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/asoc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ASoC: wm8753: fix mapping when MONOMIX is set to Stereo
ASoC: some minor changes for AD1836 and AD1938 codec drivers
ASoC: DaVinci: Fixes to McASP configuration
ASoC: Blackfin I2S: fix resuming when device hasn't been used
ASoC: Blackfin I2S: add lost platform_device parameter to resume function
ASoC: fix typos in Blackfin headers
ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long
ASoC: Blackfin AC97: add a few missing multichannel define handling
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since the active field of the dai already tells us the stream activity,
the local counter variable is redundant and can be replaced.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When MONOMIX is set to Stereo, Left PGA was not powered on but should be.
Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue.
Signed-off-by: Phil Vandry <vandry@TZoNE.ORG>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1. delete redundant assignment to bus field in spi_driver structure
2. fix lost assignment to set_bias_level entry in ad1938 codec dai
3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836"
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Get rid of that commented usage of the now defunct MODULE_PARM macro.
Signed-off-by: Robert P. J. Day <rpjday@crashcourse.ca>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
we cannot set the sampling rate of the device, but can only read it
from the board, so we don't need the member for it.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the rmh bus is not used asynchronously, so it is safe to remove the
specific code pieces.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Simplify snd_pcm_drain() implementation and avoid unneeded array-
allocation for waitqueues. Instead, one waitqueue is used for the
first draining stream, and wait until all streams finished.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This allows subsytems to provide devtmpfs with non-default permissions
for the device node. Instead of the default mode of 0600, null, zero,
random, urandom, full, tty, ptmx now have a mode of 0666, which allows
non-privileged processes to access standard device nodes in case no
other userspace process applies the expected permissions.
This also fixes a wrong assignment in pktcdvd and a checkpatch.pl complain.
Signed-off-by: Kay Sievers <kay.sievers@vrfy.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
New machine driver for WM8580 I2S i/f on SMDK64XX.
By default SoC-Slave is set and WM8580 is configured to use it's
PLLA to generate clocks from a 12MHz crystal attached to WM8580.
[Added dependency on BROKEN since the IISv4 interface hasn't been merged
yet, fixed the PLL API usage and removed the disabling of the PLL in the
hw_free function since that'll break simultaneous playback and record
-- broonie.]
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'davinci-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/khilman/linux-davinci: (62 commits)
DaVinci: DM646x - platform changes for vpif capture and display drivers
davinci: DM355 - platform changes for vpfe capture
davinci: DM644x platform changes for vpfe capture
davinci: audio: move tlv320aic33 i2c setup into board files
DaVinci: EDMA: Adding 2 new APIs for allocating/freeing PARAMs
DaVinci: DM365: Adding entries for DM365 IRQ's
DaVinci: DM355: Adding PINMUX entries for DM355 Display
davinci: Handle pinmux conflict between mmc/sd and nor flash
davinci: Add NOR flash support for da850/omap-l138
davinci: Add NAND flash support for DA850/OMAP-L138
davinci: Add MMC/SD support for da850/omap-l138
davinci: Add platform support for da850/omap-l138 GLCD
davinci: Macro to convert GPIO signal to GPIO pin number
davinci: Audio support for DA850/OMAP-L138 EVM
davinci: Audio support for DA830 EVM
davinci: Correct the number of GPIO pins for da850/omap-l138
davinci: Configure MDIO pins for EMAC
DaVinci: DM365: Add Support for new Revision of silicon
DaVinci: DM365: Fix Compilation issue due to PINMUX entry
DaVinci: EDMA: Updating default queue handling
...
Instead of always returnig pointer to the 'audio-bus' clock,
check which clock is used to generate internal clocks and
then return it's pointer.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
McASP register settings are not correct for DSP mode of operation.
There is a channel swap initally. This patch provides fixes to
the register values for proper working.
Tested on DA830/OMAP-L137 EVM, DM6467 EVM.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the sound system hasn't been utilized yet and we suspend, then we
attempt to save/restore using state that doesn't exist. So use a global
handle instead to reconfigure properly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* fix/asoc:
ASoC: remove unused #include <linux/version.h>
ASoC: S3C lrsync function made to work with IRQs disabled.
ASoC: Fix display of stream name in DAPM debugfs
ASoC: Clean up error handling in MPC5200 DMA setup
The headphone and speaker mixer elements aren't properly set for
MSI GX620 with targa-8ch-dig quirk.
Also fixed the speaker volume control for other ALC883-targa quirks,
too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit dc7d7b830e trimmed the platform_device parameter from all of the
suspend functions, but it also accidentally removed it from the resume
function in the Blackfin I2S driver. So restore it.
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Somewhere along the line, most of SND_BF5XX_MULTICHAN_SUPPORT handling was
merged, but two places were missed (the probe/resume functions). Restore
handling of this option so it gets initialized properly.
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
CDCLK can either be an output generated by the CPU, intended for use
as the CODEC master clock, or an input (probably from the CODEC)
providing a master clock for the IIS block.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch enables tlv320aic3101 support on DM365 EVM and
it was tested on DM365 EVM rev c.
Note: this patch was created based on temp/asoc branch.
Signed-off-by: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1) Explicitly set LRCLK polarity for I2S Vs LSM/MSB modes.
2) Convert from numerical to bit-field values for BCLK selection.
3) Use proper error checking for return value from clk_get
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
s3c2412_snd_lrsync() maybe reached with IRQs disabled and if LRCLK
is dead due to improper initialization of CPU or CODEC, the system
gets stuck in the loop because jiffies may never get updated.
Implemented counter based wait mechanism for atleast the same
timeout period.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The pin setup for Dell S14 quirk is rather wrong for the latest driver.
Fixed pin 0x0a, 0x0b, 0x0d and 0x0f.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unnecessary (and buggy) init sequences left for IDT92HD83*
codecs in the previous fixes. The DACs are now dynamically connected,
thus shouldn't be set statically in init verbs. Also, the mono_nid
is detected dynamically, thus shouldn't be set staticaly, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (257 commits)
[ARM] Update mach-types
ARM: 5636/1: Move vendor enum to AMBA include
ARM: Fix pfn_valid() for sparse memory
[ARM] orion5x: Add LaCie NAS 2Big Network support
[ARM] pxa/sharpsl_pm: zaurus c3000 aka spitz: fix resume
ARM: 5686/1: at91: Correct AC97 reset line in at91sam9263ek board
ARM: 5640/1: This patch modifies the support of AC97 on the at91sam9263 ek board
ARM: 5689/1: Update default config of HP Jornada 700-series machines
ARM: 5691/1: fix cache aliasing issues between kmap() and kmap_atomic() with highmem
ARM: 5688/1: ks8695_serial: disable_irq() lockup
ARM: 5687/1: fix an oops with highmem
ARM: 5684/1: Add nuc960 platform to w90x900
ARM: 5683/1: Add nuc950 platform to w90x900
ARM: 5682/1: Add cpu.c and dev.c and modify some files of w90p910 platform
ARM: 5626/1: add suspend/resume functions to amba-pl011 serial driver
ARM: 5625/1: fix hard coded 4K resource size in amba bus detection
MMC: MMCI: convert realview MMC to use gpiolib
ARM: 5685/1: Make MMCI driver compile without gpiolib
ARM: implement highpte
ARM: Show FIQ in /proc/interrupts on CONFIG_FIQ
...
Fix up trivial conflict in arch/arm/kernel/signal.c.
It was due to the TIF_NOTIFY_RESUME addition in commit d0420c83f ("KEYS:
Extend TIF_NOTIFY_RESUME to (almost) all architectures") and follow-ups.
Add the quirk entry for HP dv6. Also add a workaround for the headphone
detection by setting hp_detect=1 beforehand. Without this, the driver
won't do auto-muting because BIOS doesn't give any HP pin but only a
line-out pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's possible that hp_detect is set even though no headphone pin is
detected. The driver issues, however, an unsol event only to hp_pins[0],
which can be invalid.
This patch adds the check of the valid pin to send an unsol event
at initialization and resume callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IDT92HD73xx and STAC927x codecs use GPIO0 bit as EAPD on many machines.
However, currently we don't set it unless the model is specified just
for safety reason. But, most machines do need this bit, so this safety
handling is rather annoying.
This patch enables GPIO0 setup as default for them. Many HP / Dell
laptops should work even without model override with this change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The patch adds an interface to set the relationship between audio
channel number and slot number. The interface should be really useful
because audio channel n doesn't always use slot n in all platforms. And
for some devices, the relationship even can change with sound mode
switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Error handling code following a kzalloc should free the allocated data.
Error handling code following an ioremap should iounmap the allocated data.
The semantic match that finds the first problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,f1,l;
position p1,p2;
expression *ptr != NULL;
@@
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
<... when != x
when != if (...) { <+...x...+> }
(
x->f1 = E
|
(x->f1 == NULL || ...)
|
f(...,x->f1,...)
)
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/usb-audio:
ALSA: usb-audio - Fix types taken in min()
sound: usb-audio: do not make URBs longer than sync packet interval
sound: usb-audio: add MIDI drain callback
sound: usb-audio: use multiple output URBs
sound: usb-audio: use multiple input URBs
sound: usb-audio: Xonar U1 digital output support
* topic/tlv-minmax:
ALSA: usb-audio - Correct bogus volume dB information
ALSA: usb-audio - Use the new TLV_DB_MINMAX type
ALSA: Add new TLV types for dBwith min/max
* topic/soundcore-preclaim:
sound: make OSS device number claiming optional and schedule its removal
sound: request char-major-* module aliases for missing OSS devices
chrdev: implement __[un]register_chrdev()
* topic/oss:
ALSA: allocation may fail in snd_pcm_oss_change_params()
sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
sound: fix OSS MIDI output data loss
* topic/hda: (92 commits)
ALSA: hda - Use auto model for HP laptops with ALC268 codec
ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
ALSA: hda - Add support of Alienware M17x laptop
ALSA: hda - Remove dead codes from patch_sigmatel.c
ALSA: hda - Fix input source selection of IDT92HD73xx
ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
ALSA: hda - Unmute docking line-out as default with AD1984A codec
ALSA: hda - Add another entry for Nvidia HDMI device
ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
ALSA: hda - Fix ALC268/ALC269 headphone pin routing
ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
ALSA: hda - Add more quirk for HP laptops with AD1984A
ALSA: hda - Add / fix model entries for HD-audio driver
ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
ALSA: hda - Improve auto-cfg mixer name for ALC662
ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
ALSA: hda - Improve auto-cfg mixer name for ALC262
ALSA: hda - Improve auto-cfg mixer name for ALC260
ALSA: hda - Improve auto-cfg mixer name for ALC880
...
* topic/asoc: (226 commits)
ASoC: au1x: PSC-AC97 bugfixes
ASoC: Fix WM835x Out4 capture enumeration
ASoC: Remove unuused hw_read_t
ASoC: fix pxa2xx-ac97.c breakage
ASoC: Fully specify DC servo bits to update in wm_hubs
ASoC: Debugged improper setting of PLL fields in WM8580 driver
ASoC: new board driver to connect bfin-5xx with ad1836 codec
ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
ASoC: davinci: i2c device creation moved into board files
ASoC: Don't reconfigure WM8350 FLL if not needed
ASoC: Fix s3c-i2s-v2 build
ASoC: Make platform data optional for TLV320AIC3x
ASoC: Add S3C24xx dependencies for Simtec machines
ASoC: SDP3430: Fix TWL GPIO6 pin mux request
ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
OMAP: McBSP: Use textual values in DMA operating mode sysfs files
ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
ASoC: Select core DMA when building for S3C64xx
...
The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier,
Receiver-Amplifier and Headphone-Amplifier.
The datasheet for the ak4671 can find at the following url:
http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some chips with complex internal supply (particularly clocking)
arragements may have multiple options for some of the supply
connections. Since these don't affect user-visible audio routing
the expectation would be that they would be managed automatically
by one of the drivers.
Support these users by allowing routes to have a connected function
which is queried before the connectedness of the path is checked as
normal. Currently this is only done for supplies, other widgets
could be supported but are not currently since the expectation for
them is that audio routing will be under the control of userspace.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch fixes the following bugs:
- only reprogram bitdepth if it has changed since last call to hw_params.
- add locking inside ac97_read/write functions:
When reprogramming sample depth, the ac97 unit has to be disabled,
which should not be done in the middle of codec register accesses.
- retry timed-out codec register accesses.
- wait for status bits to set/clear when starting/stopping various
functional blocks; very important after reenabling AC97 unit else
sound may be distorted (e.g. high-pitch noise in 1kHz sine wave).
- clear fifos before/after starting/stopping RX/TX.
- longer timeouts waiting for PSC/AC97 ready after cold reset
with certain codecs this can take ridiculous amounts of time.
Run-tested on various Au1200 platforms with various codecs.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Increase the limit of PCM substreams to 128. The default value is
unchanged; only the max accept value is increased.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the debug proc file to see or change the snd_pcm_hardware fields
to emulate. The parameters can be changed by writing to a proc file like:
# echo periods_min 4 > /proc/asound/card1/dummy_pcm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add appropriate const prefix to char * arguments in proc helper functions.
Also fixed the caller side to be proper const pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-export snd_pcm_format_name() function to be used outside the PCM core.
As a first example, usbaudio is changed to use it now again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP laptops with ALC268 codec seem working better with model=auto
than model=toshiba; e.g. the auto model fixes missing digital outputs.
Let's fix quirk entry to choose auto model explicitly.
Tested-by: Jens Jorgensen <jbj1@ultraemail.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix minimum period size for cs46xx cards. This fixes a problem in the
case where neither a period size nor a buffer size is passed to ALSA;
this is the case in Audacious, OpenAL, and others.
Signed-off-by: Sophie Hamilton <kernel@theblob.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's the 8th enum of a zero indexed array. This is why I don't let
new drivers use these arrays of enums...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the volume is changed continuously (e.g., when the user drags a
volume slider with the mouse), the driver does lots of I2C writes.
Apparently, the sound chip can get confused when we poll the I2C status
register too much, and fails to complete a read from it. On the PCI-E
models, the PCI-E/PCI bridge gets upset by this and generates a machine
check exception.
To avoid this, this patch replaces the polling with an unconditional
wait that is guaranteed to be long enough.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Johann Messner <johann.messner at jku.at>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch is for the AK4671 codec driver using this format.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Instead of allocating the real buffers, use a fake buffer and ignore
read/write in the dummy driver so that we can save the resources.
For mmap, a single page (unique to the direction, though) is reused
to all buffers.
When the app requires to read/write the real buffers, pass fake_buffer=0
module option at loading time. This will get back to the old behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model clevo-m540r was created with 6-channel and digital support. All
functions verified except spdif. Tested with a VIT D2000 laptop which has:
[lspci extract]
Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller [8086:284b] (rev 03)
Subsystem: CLEVO/KAPOK Computer Device [1558:5409]
[/proc/asound/card0/codec\#0 header]
Codec: Realtek ALC883
Address: 0
Function Id: 0x1
Vendor Id: 0x10ec0883
Subsystem Id: 0x15585409
Revision Id: 0x100002
[Added a comment about HP mute and the model description by tiwai]
Signed-off-by: Dhionel Diaz <ddiaz@cenditel.gob.ve>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
More and more devices feature PLLs and FLLs with the ability to select
between multiple input clocks. In order to better support these devices
a new argument, source, has been added to the set_pll() configuration
API. Using set_clkdiv() is often difficult due to the need to stop the
PLL/FLL before any reconfiguration can be done.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Today's linux-next fails to build with
sound/arm/pxa2xx-ac97.c: In function 'pxa2xx_ac97_probe':
sound/arm/pxa2xx-ac97.c:211: error: 'pxa2xx_audio_ops_t' has no member named 'codec_data'
make[2]: *** [sound/arm/pxa2xx-ac97.o] Error 1
It looks like commit e2365bf313 has
introduced this; patch below.
Signed-off-by: Robert Schwebel <r.schwebel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the expire-time calculation in the systimer mode when the buffer
size isn't aligned to the period size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the system-timer mode, snd-dummy driver issues each tick to update
the position. This is highly inefficient and even inaccurate if the
timer can't be triggered at each tick.
Now rewritten to wake up only at the period boundary. The position
is calculated from the current jiffies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow snd-dummy driver to use high-res timer as its timing source
instead of the system timer. The new module option "hrtimer" is added
to turn on/off the high-res timer support. It can be switched even
dynamically via sysfs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The card model detection code introduced in 2.6.30 that tries to work
around partially broken EEPROM contents by reading the EEPROM directly
does not handle cards where the EEPROM has been omitted. In this case,
we have to use the default ID to allow the driver to load.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Ozan Çağlayan <ozan@pardus.org.tr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the quirk for Alienware M17x with IDT 92HD73* codec chip.
It has two HP and one line-out jack, one mic jack, a built-in
speaker and a built-in mic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the previous fix of input source for IDT92HD73xx, the amp mux
and amp vol stuff became unused. Let's rip off dead codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mux_nids to select directly the input source instead of mux
mixers so that it works with the current mux enum handler for IDT
92HD73xx codecs.
Also, clean up useless / unnecessary mixer controls and init verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bug was caught while trying to use WM8580 as I2S master on SMDK.
Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved
by referring to WM8580A manual and setting mask value correctly and
making the code to not touch 'reserved' bits of PLL4 register.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As discussed, the patch uses the original TDM order without rewriting.
For the match between TDM slot number and audio channel number, a new
API need be added.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unmute the docking-station line-out as default on machines with
AD1984A codec chip. It can be still muted via "Dock" mixer switch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not forget to program the MCLK ratio for the I2S output.
Otherwise, the master clock frequency can be too high for
the DACs at sample frequencies above 96 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocation may fail, show if it did.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
[Additional fix for invalid runtime->oss.prepare flag set by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of automatic mute and mic-switching of the docking
station HP and mic plugs for AD1984A laptop model for some HP machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the headphone pin routing of ALC268/ALC269 codecs. Using alc882
routine doesn't work because alc268/alc269 parser assumes the
independent DACs for both HP and speaker outputs. Need to assign the
DAC depending on the pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the wrong headphone output routing for MacBookPro 3,1/4,1
quirk with ALC889A codec, which caused the silent headphone output.
Also, this gives the individual Headphone and Speaker volume controls.
Reference: kernel bug#14078
http://bugzilla.kernel.org/show_bug.cgi?id=14078
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
In patch_vt1708(), the check of MUX nids is missing and this results in
the -EINVAL error in accessing Input Source mixer element. Simpliy
adding the call of get_mux_nids() fixes the problem.
Reference: Novell bnc#534904
https://bugzilla.novell.com/show_bug.cgi?id=534904
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the digital mic capture volume wasn't created. This is because
IDT codecs have output amps for digital mics, not input amps, while
input amps should be used for other analog pins. Thus the automatic
capture volume creation should check both directions for digital mics.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have
specifications to check it) have additional CLKR and FSR pins for McBSP1
receiver. Reset default is that receiver is using bit clock and frame
sync signal from those pins but it is possible to configure to use
also CLKX and FSX pins as well. In fact, other McBSP ports are doing that
internally that transmitter and receiver share the CLKX and FSX.
Add functionaly that machine drivers can set the CLKR and FSR sources by
using the snd_soc_dai_set_sysclk.
Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also, the codec setup data structure has to remain for successful
probe.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When modules are built with M= option, they pass long file paths to
__FILE__. This results in ugly outputs of snd_print*() when
CONFIG_SND_VERBOSE_PRINTK is set.
This patch adds a check of the path and strips the leading path dirs
if the file name is an absolute path to improve the readability of logs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the requested FLL configuration is the one we're currently running
in it's at best pointless to reconfigure the FLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we don't need the I2C address for the device the platform data
is redundant so allow it to be omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Chaithrika U S <chaithrika@ti.com>
No point in building them for S3C64xx, certainly no sense in running
into build issues there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since !LI_CCFG_* evaluates to 0, this did not change anything to
cfgval and ctlval.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- restructure to support multiple channel controllers by using
additional struct resources for each CC
- interface changes visible to EDMA clients
Introduce macros to build IDs from controller and channel number,
and to extract them. Modify the edma_alloc_slot function to take an
extra argument for the controller.
Also update ASoC drivers to use API. ASoC changes
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Move queue related mappings to dm<soc>.c
EDMA in DM355 and DM644x has two transfer controllers while DM646x
has four transfer controllers. Moving the queue to tc mapping and
queue priority mapping to dm<soc>.c will be helpful to probe these
mappings from platform device so that the machine_is_* testing will
be avoided.
- add channel mapping logic
Channel mapping logic is introduced in dm646x EDMA. This implies
that there is no fixed association for a channel number to a
parameter entry number. In other words, using the DMA channel
mapping registers (DCHMAPn), a PaRAM entry can be mapped to any
channel. While in the case of dm644x and dm355 there is a fixed
mapping between the EDMA channel and Param entry number.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Reviewed-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Fix the write to PMBR1 register through I2C. Also, the constant which
holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This
name is based on TRM to avoid confusion.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1) Added support of internal subwoofer (it sounds!!!)
2) Auto muting front speakers and internal subwoofer on headphones plug.
3) Internal mic works.
4) 3 channel mods (jack maps):
black pink blue
2ch: front ext mic line in
4ch: front ext mic surround
6ch: front CLFE surround
Can be changed in mixer.
5) Sound can be recorded from:
Internal mic
Ext mic
Cd
Line in
6) 2 separate capture channels.
Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more patch to give a better name for the primary output controls,
this time for ALC861-VD codec. The change is simple, just checking the
pin connection whether it's a speaker-out. When both speaker and HP
are assigned, we name the volume as "PCM" as this influences on both
outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Similar improvements for ALC262 codec like previous two commits:
assign a better name, either Master or Speaker, for the primary output
controls.
However, in the case of ALC262 codec, the necessary changes are larger
than others because we need to check the possibility of different mixer
amps depending on the pins. The pin 0x16 is mono, and bound with the
dedicated mixer 0x0e while other pins are bound with 0x0c. Thus, there
are two possible volumes.
When only one of them is used, we can name it as "Master". OTOH, when
both are used at the same time, they have to be named uniquely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of fixed "Front" mixer name, try to assign a better name, e.g.
"Master" or "Speaker" fot the primary output volume controls of ALC260
codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When there is only one DAC is used for ALC880, try to assign a better
name, either Speaker or Front, depending on the output pin type.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
s3c24xx dma has the auto reload feature, when the the trnasfer is done,
CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA
ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So
the transmission is repeated.
IRQ is issued while auto reload occurs. We change the DISRC and
DCON[19:0] in the ISR, but at this time, the auto reload has been
performed already. The first block is being re-transmitted by the DMA.
So we need rewrite the DISRC and DCON[19:0] for the next block
immediatly after the this block has been started to be transported.
The function s3c2410_dma_started() is for this perpose, which is called
in the form of "s3c2410_dma_ctrl(prtd->params->channel,
S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger().
But it is not correct. DMA transmission won't start until DMA REQ signal
arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1)
is called in s3c24xx_i2s_trigger().
In the current framework, s3c24xx_pcm_trigger() is always called before
s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in
s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or
s3c24xx_snd_rxctrl(1) is called in this function.
However, s3c2410_dma_started() is dma related, to call this function we
should provide the channel number, which is given by
substream->runtime->private_data->params->channel. The private_data
points to a struct s3c24xx_runtime_data object, which is define in
s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c
Fix this by moving the call to signal the DMA started to the DAI
drivers.
Signed-off-by: Shine Liu <liuxian@redflag-linux.com>
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a standard parser for input pins to create the input mixer
and input source controls instead of having a difference one for each
Realtek codec. The new helper parses the codec connections dynamically
isntead of fixed indicies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Functionality of functions omap_mcbsp_xmit_enable and omap_mcbsp_recv_enable
can be merged into omap_mcbsp_start and omap_mcbsp_stop since API of
those omap_mcbsp_start and omap_mcbsp_stop was changed recently allowing
to start and stop individually the transmitter and receiver.
This cleans up the code in arch/arm/plat-omap/mcbsp.c and in
sound/soc/omap/omap-mcbsp.c which was the only user for those removed
functions.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit ca6e2ce086 is setting up few XCCR and
RCCR bits for I2S and DPS_A formats. Part of the bits are already set
for all formats and I believe that XDISABLE and RDISABLE bits are
format independent.
As XCCR and RCCR are found only from OMAP2430 and OMAP34xx, I move setup
of XDISABLE and RDISABLE to where those cpu's are tested and remove format
dependent part for simplicity.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_interval_list() expected a sorted list but did not document this, so
there are drivers that give it an unsorted list. To fix this, change
the algorithm to work with any list.
This fixes the "Slave PCM not usable" error with USB devices that have
multiple alternate settings with sample rates in decreasing order, such
as the Philips Askey VC010 WebCam.
http://bugzilla.kernel.org/show_bug.cgi?id=14028
Reported-and-tested-by: Andrzej <adkadk@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reuse a part of the code of ALC268 parser for ALC269.
This will change the default output volume either to Front or Speaker
depending on the pin configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a logic error in the range check of the input level control that
would prevent setting any volume less than the maximum.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify loops in such way that the register value is checked also after
the timeout condition, just in case the heavy interrupt load etc. caused
the thread to sleep for the time period exceeding the timeout value.
While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready().
Reported-by: Jack Byer <ojbyer@usa.net>
Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Free socdev if snd_soc_init_card() fails.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.
Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.
In addition to the previously displayed information active streams
are also shown in these files.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is very simple driver for ALSA
It supprt headphone output and stereo input only
This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the tlv320aic3x CODEC on the Simtec Hermes board.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixup the device changes by modifying the files that we just removed the
explicit device creation from with i2c_register_board_info() until this
can be moved into the relevant board files.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tlv320aic3x driver managed its own i2c device, instead of an extant
one created by the board support code. Change the code to make it so that
the driver binds to an extant (in this case i2c) device.
Add explict tlv320aic33 as well as tlv320aic3x to the supported device
table and remove the old driver bindings from the users of this code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add core support for the range of S3C24XX Simtec boards with TLV320AIC23
CODECs on them. Since there are also boards with similar IIS routing the
AMP and the configuration code is placed in a core file for re-use with
other CODEC bindings.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Configures DMA sync mode depending on McBSP operating mode value.
The value is configurable by McBSP instance. So, depending
on McBSP operating mode, the DMA sync mode is passed from
omap-mcbsp to omap-pcm. Besides that, it also configures
McBSP threshold value depending on which McBSP mode is activated.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch changes the way DMA is done in omap-pcm.c
in order to reduce power consumption. There is no need
to have so much SW control in order to have DMA in idle
state during audio streaming. Configuring McBSP threshold value
and DMA to FRAME_SYNC are sufficient.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All these steps are required for ASoC to behave correctly.
rccr and xccr are format dependent, for example TDM audio
has different values than I2S or DSP_A. Also the
omap_mcbsp_xmit_enable and/or omap_mcbsp_recv_enable must
be called right after the DMA has started.
This provides no longer L and R channels switching at random.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is, no RFIG or XFIG (Not defined in 34xx), correct
initiliazation of rccr and xccr.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Align DMA address to DMA burst transaction sizes.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Improve DMA transfers by enabling Burst transaction.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver is very simple.
It support playback only now.
This patch is tested by ms7724se board.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> Then it's a driver bug. If unaligned period size is allowed, it means
> that the irq is really generated in that period, not at the buffer
> boundary. Otherwise, it must have a proper hw-constraint to align the
> period size to the buffer size.
This patch will fix the bug metioned in the above mail. Force the peroid
size to be aligned with the buffer size.
Based and tested on linux-2.6.31-rc6.
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Peter Ujfalusi