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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>
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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>
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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>
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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>
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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>
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More entries for HP laptops to get them working properly.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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We now need the PCM header to kick the DMA.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
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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>
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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>
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- 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>
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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>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/misc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
sound: pcm_lib: fix unsorted list constraint handling
sound: vx222: fix input level control range check
ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
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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>
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The last patch in this series is for ALC662; pretty similar as the
previous patch for ALC861-VD.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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Commit ca6e2ce08679c094878d7f39a0349a7db1d13675 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>
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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>
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Ensure that the core DMA support is available when building for
S3C64xx.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
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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>
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Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Takashi Iwai <tiwai@suse.de>
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To fix compile errors.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Remove code covered by #if/endif 0 and #ifdef/endif CODEC_RESET
(CODEC_RESET is never defined).
Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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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>
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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>
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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>
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This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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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>
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