Commit Graph

6132 Commits

Author SHA1 Message Date
Takashi Iwai a68c4d1133 ALSA: dummy - Fake buffer allocations
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>
2009-09-07 09:01:10 +02:00
ddiaz@cenditel.gob.ve a65cc60f63 ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
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>
2009-09-07 07:32:33 +02:00
Linus Torvalds b71b7dc09a Merge branch 'fix/oxygen' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/oxygen' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  sound: oxygen: handle cards with missing EEPROM
  sound: oxygen: fix MCLK rate for 192 kHz playback
2009-09-05 14:55:30 -07:00
Robert Schwebel 367da1527a ASoC: fix pxa2xx-ac97.c breakage
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>
2009-09-04 20:19:56 +01:00
Takashi Iwai b5d1078173 ALSA: dummy - Fix the timer calculation in systimer mode
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>
2009-09-04 08:45:11 +02:00
Takashi Iwai b142037b4c ALSA: dummy - Better jiffies handling
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>
2009-09-03 16:01:06 +02:00
Takashi Iwai c631d03c68 ALSA: dummy - Support high-res timer mode
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>
2009-09-03 15:59:26 +02:00
Clemens Ladisch 92653453c3 sound: oxygen: handle cards with missing EEPROM
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>
2009-09-03 07:38:06 +02:00
Mark Brown 2eff31e809 ASoC: Fully specify DC servo bits to update in wm_hubs
Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-09-02 19:36:22 +01:00
Takashi Iwai 842ae63800 ALSA: hda - Add support of Alienware M17x laptop
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>
2009-09-02 07:43:08 +02:00
Takashi Iwai 4a9678909b ALSA: hda - Remove dead codes from patch_sigmatel.c
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>
2009-09-02 01:09:54 +02:00
Takashi Iwai e2aec17100 ALSA: hda - Fix input source selection of IDT92HD73xx
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>
2009-09-02 01:00:05 +02:00
Takashi Iwai d94ff6b7ca ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
Fix the old dead CONFIG_SND_DEBUG_DETECT to CONFIG_SND_DEBUG_VERBOSE.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-02 00:20:21 +02:00
jassi brar 5c0d38c947 ASoC: Debugged improper setting of PLL fields in WM8580 driver
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>
2009-09-01 11:37:41 +01:00
Barry Song dce944dbb2 ASoC: new board driver to connect bfin-5xx with ad1836 codec
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>
2009-09-01 11:36:13 +01:00
Takashi Iwai 2ad81ba014 ALSA: hda - Unmute docking line-out as default with AD1984A codec
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>
2009-09-01 09:09:26 +02:00
Takashi Iwai f8ff035e38 ALSA: hda - Add another entry for Nvidia HDMI device
Added another entry for Nvidia HDMI device (10de:0003).

Reference: kernel bug#14097
	http://bugzilla.kernel.org/show_bug.cgi?id=14097

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-01 08:53:19 +02:00
Clemens Ladisch b91ab72b83 sound: oxygen: fix MCLK rate for 192 kHz playback
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>
2009-09-01 08:45:40 +02:00
Linus Torvalds cda9856f1c Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix MacBookPro 3,1/4,1 quirk with ALC889A
  ALSA: hda - Add missing mux check for VT1708
2009-08-31 17:36:10 -10:00
Roel Kluin cbbb05703d ALSA: allocation may fail in snd_pcm_oss_change_params()
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>
2009-08-31 16:33:23 +02:00
Takashi Iwai fe7e56814c ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
A similar initialization of GPIO1 pin like mobile model is needed
for laptop model, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-31 08:37:46 +02:00
Takashi Iwai 17bbaa6f60 ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
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>
2009-08-31 08:32:27 +02:00
Takashi Iwai be0ae923a4 Merge branch 'fix/hda' into topic/hda
Conflicts:
	sound/pci/hda/patch_realtek.c
2009-08-31 08:27:10 +02:00
Takashi Iwai e9af4f365f ALSA: hda - Fix ALC268/ALC269 headphone pin routing
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>
2009-08-31 08:25:58 +02:00
Takashi Iwai a3f730af7e ALSA: hda - Fix MacBookPro 3,1/4,1 quirk with ALC889A
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>
2009-08-31 08:23:13 +02:00
Takashi Iwai 0f67a61162 ALSA: hda - Add missing mux check for VT1708
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>
2009-08-31 08:12:29 +02:00
Takashi Iwai 96f845de89 ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
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>
2009-08-29 00:49:36 +02:00
Jarkko Nikula d2c0bdaa93 ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
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>
2009-08-28 18:36:43 +01:00
Chaithrika U S f4890b5c04 ASoC: davinci: i2c device creation moved into board files
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>
2009-08-28 10:33:10 +01:00
Takashi Iwai 36ce99c1dc ALSA: Add debug module option
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>
2009-08-27 17:42:08 +02:00
Takashi Iwai 286f5875ca ALSA: hda - Add more quirk for HP laptops with AD1984A
More entries for HP laptops to get them working properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-27 14:37:51 +02:00
Takashi Iwai 1b0053a0f0 ALSA: core - strip too long file names in snd_print*()
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>
2009-08-27 12:39:35 +02:00
Mark Brown f1e887de2d ASoC: Don't reconfigure WM8350 FLL if not needed
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>
2009-08-26 15:27:57 +01:00
Mark Brown 5dc0748182 ASoC: Fix s3c-i2s-v2 build
We now need the PCM header to kick the DMA.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-26 15:27:57 +01:00
Mark Brown 977d49e00d ASoC: Make platform data optional for TLV320AIC3x
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>
2009-08-26 15:27:56 +01:00
Mark Brown bc36681fdc ASoC: Add S3C24xx dependencies for Simtec machines
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>
2009-08-26 15:27:56 +01:00
Roel Kluin f1d269bac2 sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
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>
2009-08-26 12:42:43 +02:00
Sudhakar Rajashekhara 60902a2cb1 davinci: EDMA: multiple CCs, channel mapping and API changes
- 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>
2009-08-26 10:56:56 +03:00
Candelaria Villareal, Jorge 30cd0c4ad5 ASoC: SDP3430: Fix TWL GPIO6 pin mux request
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>
2009-08-25 19:30:32 +01:00
Linus Torvalds a206e9417f Merge branch 'fix/misc' of 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()
2009-08-25 09:47:06 -07:00
Denis Kuplyakov fc86f95415 ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
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>
2009-08-25 18:16:55 +02:00
Takashi Iwai 0d884cb936 ALSA: hda - Improve auto-cfg mixer name for ALC662
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>
2009-08-25 16:14:35 +02:00
Takashi Iwai a4fcd49109 ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
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>
2009-08-25 16:12:15 +02:00
Takashi Iwai c3fc1f502a ALSA: hda - Improve auto-cfg mixer name for ALC262
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>
2009-08-25 16:08:47 +02:00
Takashi Iwai 23112d6d2d ALSA: hda - Improve auto-cfg mixer name for ALC260
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>
2009-08-25 16:07:08 +02:00
Takashi Iwai cb162b6bf2 ALSA: hda - Improve auto-cfg mixer name for ALC880
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>
2009-08-25 16:05:03 +02:00
Shine Liu faf907c7ba ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
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>
2009-08-25 13:09:05 +01:00
Takashi Iwai 05f5f47708 ALSA: hda - Generalize input pin parsing in patch_realtek.c
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>
2009-08-25 13:10:18 +02:00
Jarkko Nikula d09a2afc93 ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
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>
2009-08-25 10:20:48 +01:00
Jarkko Nikula 32080af7a6 ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
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>
2009-08-25 10:20:48 +01:00
Clemens Ladisch b1ddaf681e sound: pcm_lib: fix unsorted list constraint handling
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>
2009-08-25 08:52:34 +02:00
Mark Brown e4aa8dd5ca Merge branch 'topic/digital-mixing' into for-2.6.32 2009-08-24 20:44:41 +01:00
Mark Brown 239a22aaa9 ASoC: Select core DMA when building for S3C64xx
Ensure that the core DMA support is available when building for
S3C64xx.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-24 20:42:48 +01:00
Takashi Iwai 9d0b71b1cf ALSA: hda - Reuse ALC268 parser for ALC269
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>
2009-08-24 14:10:30 +02:00
Clemens Ladisch edd1365e90 sound: vx222: fix input level control range check
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>
2009-08-24 09:46:08 +02:00
Wu Fengguang fd72d00846 ALSA: hda: move open coded tricks into get_wcaps_channels()
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 09:42:48 +02:00
Takashi Iwai c6ea2af76a ASoC: Remove unneeded inclusion of linux/regulator/consumer.h
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 09:41:32 +02:00
Takashi Iwai 20496ff378 ASoC: add missing inclusion of debugfs.h
To fix compile errors.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-24 09:41:05 +02:00
Marek Vasut e2365bf313 ASoC: Pass correct platform data from pxa2xx-ac97
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-23 18:18:01 +01:00
Bartlomiej Zolnierkiewicz 848bffef28 ALSA: ali5451: remove dead code
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>
2009-08-23 18:59:14 +02:00
Bartlomiej Zolnierkiewicz 70bdbd3d1a ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
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>
2009-08-23 18:58:07 +02:00
Roel Kluin 821ebc86ef ASoC: free socdev if init_card() fails in wm9705_soc_probe()
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>
2009-08-23 10:41:06 +01:00
Mark Brown 79fb9387f8 ASoC: Add DAPM widget power decision debugfs files
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>
2009-08-21 17:17:59 +01:00
Kuninori Morimoto b8e583f601 ASoC: Add FSI-AK4642 sound support for SuperH
This patch is tested by ms7724se

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-21 11:02:03 +01:00
Kuninori Morimoto a3a83d9a7c ASoC: Add ak4642/ak4643 codec support
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>
2009-08-21 10:54:02 +01:00
Ben Dooks b2ec22e263 ASoC: S3C24XX: Support for Simtec Hermes boards
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>
2009-08-21 10:53:06 +01:00
Ben Dooks aa6b904e66 ASoC: tlv320aic3x: fixup board device changes
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>
2009-08-21 10:52:57 +01:00
Ben Dooks cb3826f524 ASoC: tlv320aic3x: Change to use device model
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>
2009-08-21 10:52:49 +01:00
Ben Dooks 14412acde5 ASoC: S3C24XX: Add audio core and tlv320aic23 for Simtec boards
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>
2009-08-21 10:52:42 +01:00
Eduardo Valentin a0a499c579 ASoC: OMAP: Use DMA operating mode of McBSP
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>
2009-08-20 20:10:29 +01:00
Eduardo Valentin caebc0cb3b ASoC: OMAP: Use McBSP threshold to playback and capture
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>
2009-08-20 20:10:29 +01:00
Eero Nurkkala ca6e2ce086 ASoC: Always syncronize audio transfers on frames
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>
2009-08-20 20:10:29 +01:00
Eero Nurkkala c721bbdad7 ASoC: Add runtime check for RFIG and XFIG
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>
2009-08-20 20:10:28 +01:00
Eduardo Valentin a152ff24b9 ASoC: OMAP: Make DMA 64 aligned
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>
2009-08-20 20:10:28 +01:00
Eduardo Valentin 9599d485cb ASoC: OMAP: Enable DMA burst mode
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>
2009-08-20 20:10:28 +01:00
Kuninori Morimoto a4d7d550a9 ASoC: Add SuperH FSI driver support for ALSA
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>
2009-08-20 20:01:42 +01:00
Shine Liu f61c890ec6 ASoC: S3C24XX : Align the peroid size to the buffer size
> 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>
2009-08-20 19:42:40 +01:00
Linus Torvalds a1d1251115 Merge branch 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec
  ALSA: hda: add model for Intel DG45ID/DG45FC boards
  ALSA: hda: enable speaker output for Compaq 6530s/6531s
2009-08-20 10:19:39 -07:00
Takashi Iwai 4cdc115fd3 ALSA: pcm - Fix drain behavior in non-blocking mode
The current PCM core has the following problems regarding PCM draining
in non-blocking mode:

- the current f_flags isn't checked in snd_pcm_drain(), thus changing
  the mode dynamically via snd_pcm_nonblock() after open doesn't work.
- calling drain in non-blocking mode just return -EAGAIN error, but
  doesn't provide any way to sync with draining.

This patch fixes these issues.
- check file->f_flags in snd_pcm_drain() properly
- when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state
  but quits ioctl immediately without waiting the whole drain; the
  caller can sync the drain manually via poll()

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-20 16:40:16 +02:00
Mark Brown f8bae4caaa ALSA: Restore support for DMAless DAIs on PXA
Used for applications such as direct bluetooth connections on
smartphones which don't go via the CPU. This used to be supported
before the refactoring to share code but this check was removed
during that move.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-19 20:30:14 +01:00
Takashi Iwai 454e134d0e Merge branch 'fix/hda' into topic/hda 2009-08-19 20:10:24 +02:00
Takashi Iwai 3abf2f3639 ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec
There are many variants of Toshiba laptops with ALC268 codec, and
it seems that a few of them don't work with model=toshiba preset
since they have the secondary ALC268 codec just for HDMI output.
This is a regression due to the previous clean-up work to merge all
Toshiba quirk entries into a single check.

This patch adds the identification of such laptops to apply the
standard BIOS-probing method.  Unfortunately, Toshiba laptops have
all the same PCI SSID, so we need to check the codec SSID to identify
each device.

Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 20:05:02 +02:00
Mark Brown 474e09ca01 ASoC: Provide default set_bias_level() implementation
If the CODEC does not provide a set_bias_level() then update the
bias_level variable for it since other parts of the system expect
that to be maintained.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-19 14:18:53 +01:00
Takashi Iwai 1c11ce8118 Merge branch 'fix/hda' into topic/hda 2009-08-19 12:11:06 +02:00
Wu Fengguang ae709440ed ALSA: hda: add model for Intel DG45ID/DG45FC boards
The BIOS pin configs are in fact correct and shall not be overwritten.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 12:10:25 +02:00
Wu Fengguang 150fe14c1a ALSA: hda: enable speaker output for Compaq 6530s/6531s
HP Compaq 6530s and 6531s internal speaker is silence or becomes silence
within 1 minute after fresh boot. It is found that pin 0x1c must be set to
PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and
speaker pin 0x16 seem to be unrelated.

The codec differences before/after patch are:

@@ Node 0x17 [Pin Complex] wcaps 0x40020b:
   Pin Default 0x41a6e130: [N/A] Mic at Ext Rear
     Conn = Digital, Color = White
     DefAssociation = 0x3, Sequence = 0x0
     Misc = NO_PRESENCE
-  Pin-ctls: 0x24: IN
+  Pin-ctls: 0x40: OUT
@@ Node 0x1c [Pin Complex] wcaps 0x40018d:
   Pin Default 0x41813021: [N/A] Line In at Ext Rear
     Conn = 1/8, Color = Blue
     DefAssociation = 0x2, Sequence = 0x1
-  Pin-ctls: 0x24: IN VREF_80
+  Pin-ctls: 0x40: OUT VREF_HIZ
   Unsolicited: tag=00, enabled=0
   Connection: 1
      0x24

Tests show that it won't impact (external) Mic recording.

Reported-by: "Lin, Ming M" <ming.m.lin@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 12:07:27 +02:00
Takashi Iwai fdbc66266c ALSA: hda - Fix invalid capture mixers with some ALC268 models
The auto-mic clean-up patches caused regressions on some ALC268 models
that have no proper input_mux but with "Input Source" mixer elements.
Such a combination results in Oops when accessed.

[A reason why set_capture_mixer() isn't used in patch_alc268() is that
ALC268 codec have HDA_OUTPUT direction for capture volumes unlike other
codecs.  Thus it needs own definitions of capture elements.]

This patch fixes the issues:
- Add a capture mixer definition without input-source
- Use the new capture mixer appropriately

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-19 00:22:17 +02:00
Mark Brown 59ae07a580 ASoC: WM8993 digital mixing support
The WM8993 provides digital sidetone paths and also allows each
channel on the audio interface to be routed separtately to the
DACs and ADCs.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:06:13 +01:00
Mark Brown 010ff26226 ASoC: Add input and output AIF widgets
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.

To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.

A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:06:08 +01:00
Mark Brown d1a5e44b89 ASoC: Remove duplicate ADC/DAC widgets from wm_hubs.c
These need to be in the CODEC since the DAIs supported by the CODECs
aren't static.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:04:24 +01:00
Mark Brown b2472b1d4c ASoC: Reenable S3C64xx I2S support
Joonyoung Shim reports that S3C64xx I2S is working on the NCP boards so
allow it to be selected in Kconfig.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmciro.com>
2009-08-18 16:02:59 +01:00
Joonyoung Shim 0914b93f4f ASoC: Fix data format configuration for S3C64XX IISv2
The data format configuration for S3C64xx IISv2 was hardcoded for IISMOD
register. This patch changes to the defined values it.

And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14.

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-18 16:02:36 +01:00
Mark Brown d3c9e9a139 ASoC: Implement TDM configuration for WM8993
Note that the number of slots used internally is specified in terms
of stereo slots while the external API works with mono slots.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 18:53:50 +01:00
Mark Brown 0182dcc52c ASoC: Fix WM8993 MCLK configuration for high frequency MCLKs
When used without the PLL we were accidentally clearing the MCLK/2
divider, resulting in a double rate SYSCLK when the divider should
have been used.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 18:53:44 +01:00
Russell King 29c08460d4 Merge branch 'next-s3c' of git://aeryn.fluff.org.uk/bjdooks/linux into devel-stable 2009-08-17 18:16:28 +01:00
Mark Brown 1ca04065c3 ASoC: Power speakers and headphones simultaneously
Speaker and headphone outputs do not need to be handled separately
since they can't be part of the same path.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 16:26:59 +01:00
Mark Brown b14b76a56e ASoC: Fix handling of bias levels for non-DAPM codecs
If the system doesn't have any DAPM widgets then we can't use their
state to check if the bias level for the codec should be up.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 12:57:59 +01:00
Shine Liu 0c093fb542 ASoC: UDA134X: Fix mistaken mute/unmute code
There is a mistake in current uda134x_mute function: mute_reg has been
changed in line 162 or line 164, so uda134x_write should write
"mute_reg" but not "mute_reg & ~(1<<2)" to
UDA134X_DATA010.

Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 12:56:57 +01:00
Clemens Ladisch 18dd0aa5af sound: snd_ctl_remove_user_ctl: prevent removal of kernel controls
Ensure that userspace can remove only user controls.  Controls created
by kernel drivers must not be removed because they might be referenced
in calls to snd_ctl_notify().

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:21 +02:00
Clemens Ladisch f217ac59b6 sound: snd_ctl_remove_unlocked_id: simplify user control counting
Move the decrementing of the user controls counter from
snd_ctl_elem_remove to snd_ctl_remove_unlocked_id; this saves the
separate locking of the controls semaphore, and therefore removes
a harmless race.

Since the purpose of the function is to operate on user controls (the
control being unlocked is just a prerequisite), rename it to
snd_ctl_remove_user_ctl.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:15 +02:00
Clemens Ladisch 317b80817f sound: snd_ctl_remove_unlocked_id: simplify error paths
Use a common exit path to release the mutex and to return a possible
error.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:06 +02:00
Clemens Ladisch 2a031aedf7 sound: snd_ctl_elem_add: fix value count check
Make sure that no user element that has no values can be added.

The check for count>1024 is not needed because the count is checked
later for the individual control types.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-17 12:48:00 +02:00
Janusz Krzysztofik 471e3dec3a ASoC: OMAP: Enhance OMAP1510 DMA progress software counter
Enhance period_index accuracy, particularly just before buffer rewind, by
making use of DMA interrupt status flags in addition to simply counting up
interrupts.

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 11:00:34 +01:00
Janusz Krzysztofik 64844a6ac8 ASoC: OMAP: Make use of DMA channel self linking on OMAP1510
Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP
models. Remove unnecessary DMA transfer restart from interrupt handler
routine.

The interrupt routine used to maintain a period index, originally needed for
counting up periods up to a full buffer in order to restart the DMA transfer.
For some time, this counter is also used as a replacement for hardware DMA
progress counter that has been found unusable on OMAP1510 in case of playback.
Thus, the period index calculation cannot be omitted completely. However, the
accuracy of this counter can still suffer from missing DMA interrupts.

In order to work correctly, it requires patch 1 from this series also applied:
[RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510

Created against linux-2.6.31-rc5.

Tested on Amstrad Delta.

Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-17 10:59:59 +01:00
Mark Brown 1e97f50b70 ASoC: Factor out cache I/O from WM8974
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 12:15:10 +01:00
Mark Brown 37cfa1950e Merge branch 'wm8974-upstream' into for-2.6.32 2009-08-15 11:52:43 +01:00
Mark Brown 29e02cb3ff ASoC: Hook i.MX into build
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:37:30 +01:00
Mark Brown d555a552ae ASoC: Staticise unexported variables
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:36:49 +01:00
Mark Brown a2d512a978 ASoC: Remove unneeded i.MX dependency on SND
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-15 11:36:20 +01:00
Mark Brown 08229de4b4 Merge branch 'for-2.6.32' into mxc
Conflicts:
	sound/soc/Makefile
2009-08-15 11:20:44 +01:00
Takashi Iwai 7570ef1834 ALSA: hda - Add missing num_adc_nids definition for IDT92HD8xxx
The previous fix removed the definition of num_adc_nids wrongly, and
this resulted in the missing input-source control.  Now readded again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-15 11:57:53 +02:00
Barry Song 2a708137fd ASoC: delete -spi suffix in ad1938 and free private data while registers fail
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 17:53:02 +01:00
Peter Ujfalusi 9028935d75 ASoC: TWL4030: Fix for capture mixer strings
Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-14 17:52:59 +01:00
Mark Brown d91e9a7ab9 ARM: S3C24XX: Add platform device for AC97 controller
Move the definition of the "generic" IRQ in the process.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
2009-08-14 01:13:29 +01:00
Marek Vasut 4ac0478f2a ALSA: Allow passing platform_data for pxa2xx-ac97
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..

Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:37 +01:00
Chaithrika U S 30230f4cd7 ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:36 +01:00
Chaithrika U S 517ee6cf69 ASoC: DaVinci: Add a DAI format to McASP driver
The patch adds a DAI format: Codec bit clock master and frame sync slave,
to the driver.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:35 +01:00
Chaithrika U S 6a99fb5fb8 ASoC: DaVinci: McASP driver enhacements
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf

This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.

Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:35 +01:00
Mark Brown a2342ae325 ASoC: Factor out shared code from WM8993
The WM8993 analogue control is shared with other devices in the same
product line.  Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 22:19:31 +01:00
Takashi Iwai 667067d898 ALSA: hda - Fix / clean up IDT92HD83xxx codec parser
A few improvements for IDT 92HD83xxx codec pareser:
- Remove unused / deprecated mixer-amp controls
- Handle d-mics as normal inputs since this codec has no separate
  MUXes for analog and digital
- Don't create duplicated controls for capture volumes with Mux
  capture volumes

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-13 18:14:42 +02:00
Takashi Iwai a6cd7a71fd Merge branch 'topic/hda-dmic-fix' into topic/hda 2009-08-13 18:14:02 +02:00
Mark Brown e9ade7f933 ASoC: Minor cleanups to AD1938 driver
- Build in SND_SOC_ALL_CODECS.
- Remove null suspend/resume stuff.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 15:19:42 +01:00
Barry Song 7eaae41ea5 new ad1836 codec driver based on asoc
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 15:18:53 +01:00
Peter Ujfalusi 9008adf9a9 ASoC: TWL4030: Introduce PGAs for outputs
Dynamically control and control only the needed output amplifier
muting/un-muting.

The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.

Move these as separate PGA so only the needed amplifier will be touched.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 14:56:13 +01:00
Barry Song c4ff357ada ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM work
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.

Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-13 10:47:22 +01:00
Takashi Iwai 1c4bdf9be0 ALSA: hda - Enable line-out detection only with speakers
Enable line-out detection for IDT/STAC codecs only when speaker pins
exist.  In some cases, the speaker itself is identified as line-out,
and this confuses the situation.  Only the extra line-outs should do
auto-muting.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-13 08:27:38 +02:00
Tim Blechmann c18bc9b927 ALSA: hdsp - allow proc reporting with disconnected io box
the hdsp driver refuses to report any information via the proc
interface, if the io box is not connected. with this patch, the
content of the control and status registers is printed before the
iobox check.

Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-12 18:21:30 +02:00
Mark Brown d2a382143b ASoC: Update AD1938 for new TDM slot API
It's only actually paying attention to the slot count anyway.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-12 14:30:33 +01:00
Takashi Iwai 8884be98bc Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Don't override ADC definitions for ALC codecs
  ALSA: hda - Add missing vmaster initialization for ALC269
2009-08-12 08:05:20 +02:00
Takashi Iwai 909a2607a5 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Add missing DRV_NAME definitions for fsl/* drivers
2009-08-12 08:05:19 +02:00
Herton Ronaldo Krzesinski 5908589f31 ALSA: hda - fix noise issue when recording from digital mic with alc268
With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).

Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-12 07:35:02 +02:00
Takashi Iwai 4f5d170620 ALSA: hda - Clean up init and setup hooks for Realtek codecs
Move static codes to setup from init_hook for each model.

Also, use the common auto-mic selection helper for devices that support
auto-mic selection.  They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 18:17:46 +02:00
Mark Brown e0026beac0 ASoC: Update WM9081 for tdm_slot() API change
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.

Also allow 0 TDM slots to be configure (for use when disabling TDM).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-11 16:29:21 +01:00
Takashi Iwai e9c364c04f ALSA: hda - Add setup hook to ALC preset struct
Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 17:16:13 +02:00
Takashi Iwai 4d8e22e0f6 ALSA: hda - Add a white-list for MSI option
Created a white-list to enable MSI since some devices require MSI
explicitly due to BIOS/ACPI problems.  Simply using a quirk list.
As the first case, take HP Compaq CQ40.

Reference: Novell bnc#529971
	https://bugzilla.novell.com/show_bug.cgi?id=529971

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 14:25:46 +02:00
Mark Brown 1921bab217 Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32 2009-08-11 13:09:27 +01:00
Randy Dunlap 17244c24f9 ASoC: fix I2C build errors
Fix soc build errors when I2C is built as a loadable module:

(.text+0x5d26b): undefined reference to `i2c_master_send'
soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer'

Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-11 10:47:25 +01:00
Takashi Iwai b59bdf3b0c ALSA: hda - Check connectivity for auto-mic of Realtek codecs
Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC.  Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.

This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.

Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 09:47:30 +02:00
Takashi Iwai 52b5deefbb Merge branch 'fix/hda' into topic/hda 2009-08-11 08:47:38 +02:00
Takashi Iwai dd704698f5 ALSA: hda - Don't override ADC definitions for ALC codecs
ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition.  This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).

This patch avoids the overriding by adding the proper checks.

Reference: Novell bnc#529467
	https://bugzilla.novell.com/show_bug.cgi?id=529467

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 08:45:11 +02:00
Takashi Iwai f1e6d3c5cf ALSA: usb-audio - Fix types taken in min()
Fix the compile warning due to different integer types used in min():
  sound/usb/usbaudio.c: In function 'init_substream_urbs':
  sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-11 08:16:15 +02:00
Takashi Iwai 2a22d3f812 ALSA: hda - Use only one capture stream for auto-mic
When the auto-mic feature is enabled, we should support only one
capture stream.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 18:56:05 +02:00
Takashi Iwai 6c81949227 ALSA: hda - Add auto-mic support for Realtek codecs
Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode).  The auto-mic mode is enabled
only when one internal mic and one external mic are present.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 18:47:44 +02:00
Tejun Heo 93fe4483e6 sound: make OSS device number claiming optional and schedule its removal
If any OSS support is enabled, regardless of built-in or module,
sound_core claims full OSS major number (that is, the old 0-255
region) to trap open attempts and request sound modules using custom
module aliases.  This feature is redundant as chrdev already has such
mechanism.  This preemptive claiming prevents alternative OSS
implementation.

The custom module aliases are scheduled to be removed and the previous
patch made soundcore emit the standard chrdev aliases too to help
transition.

This patch schedule the feature for removal in a year and makes it
optional so that developers and distros can try new things in the
meantime without rebuilding the kernel.  The pre-claiming can be
turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter
soundcore.preclaim_oss.

As this allows sound minors to be individually grabbed by other users,
this patch updates sound_insert_unit() such that if registering
individual device region fails, it tries the next available slot.

For details on removal plan, please read the entry added by this patch
in feature-removal-schedule.txt .

Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:59:36 +02:00
Mark Brown e0c48a18f7 ASoC: Drop unneeded declaration of removed wm8731 SPI write function
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-08-10 12:59:29 +01:00
Tejun Heo 0a848680a8 sound: request char-major-* module aliases for missing OSS devices
Till now missing OSS devices emitted sound-slot/service-* module
alises instead of the standard char-major-* if a missing device number
is opened if soundcore is loaded.  The custom module aliases don't
have any inherent benefit than backward compatibility.

sound-slot/service-* module aliases is scheduled to be removed and to
help the transition this patch makes soundcore emit the standard
module alises along with the custom ones.

Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:59:26 +02:00
Clemens Ladisch 5e8e7c3853 sound: fix OSS MIDI output data loss
In the 2.1.6 kernel, the output loop in midi_poll() was changed to
enable interrupts during the outputc() call.  Unfortunately, the check
whether the device has accepted the current byte ("ok") was moved behind
the code that removes the byte from the output queue, so one byte would
be lost every time the hardware FIFO is full.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:15:43 +02:00
Clemens Ladisch 6e2efaacb3 sound: ymfpci: increase timer resolution to 96 kHz
Allow the interval timer to be programmed with its full 96 kHz
precision.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:14:46 +02:00
Clemens Ladisch 765e8db078 sound: usb-audio: do not make URBs longer than sync packet interval
Using more packets in one URB do avoid interrupts does not make sense
when we have a sync pipe whose packets generate interrupts more often.
Therefore, limit the URB size to the synchronization packet interval.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-10 13:13:56 +02:00
Takashi Iwai d5c9c8912a Merge branch 'fix/hda' into topic/hda 2009-08-10 11:58:09 +02:00