Commit graph

6107 commits

Author SHA1 Message Date
Takashi Iwai
0f23c5cc50 Merge branch 'topic/midi' into for-linus
* topic/midi:
  sound: rawmidi: disable active-sensing-on-close by default
  sound: seq_oss_midi: remove magic numbers
  sound: seq_midi: do not send MIDI reset when closing
  seq-midi: always log message on output overrun
2009-09-10 15:32:56 +02:00
Takashi Iwai
8a3351bbb9 Merge branch 'topic/ice1724-pm' into for-linus
* topic/ice1724-pm:
  ALSA: ice1724 - Fix section mismatch
  ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD2
2009-09-10 15:32:55 +02:00
Takashi Iwai
dcb37d509a Merge branch 'topic/hdsp' into for-linus
* topic/hdsp:
  ALSA: hdsp - allow proc reporting with disconnected io box
2009-09-10 15:32:54 +02:00
Takashi Iwai
2d4ff66ad7 Merge branch 'topic/hda' into for-linus
* 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
  ...
2009-09-10 15:32:52 +02:00
Takashi Iwai
6a0f402146 Merge branch 'topic/dummy' into for-linus
* topic/dummy:
  ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128
  ALSA: dummy - Add debug proc file
  ALSA: Add const prefix to proc helper functions
  ALSA: Re-export snd_pcm_format_name() function
  ALSA: dummy - Fake buffer allocations
  ALSA: dummy - Fix the timer calculation in systimer mode
  ALSA: dummy - Add more description
  ALSA: dummy - Better jiffies handling
  ALSA: dummy - Support high-res timer mode
2009-09-10 15:32:51 +02:00
Takashi Iwai
f9892a52e2 Merge branch 'topic/dma-sgbuf' into for-linus
* topic/dma-sgbuf:
  ALSA: Fix SG-buffer DMA with non-coherent architectures
2009-09-10 15:32:50 +02:00
Takashi Iwai
6c5cb93b1e Merge branch 'topic/ctxfi' into for-linus
* topic/ctxfi:
  ALSA: ctxfi - Simple code clean up
  ALSA: ctxfi - Native timer support for emu20k2
2009-09-10 15:32:48 +02:00
Takashi Iwai
f604529d0c Merge branch 'topic/ctl-add-remove-fixes' into for-linus
* topic/ctl-add-remove-fixes:
  sound: snd_ctl_remove_user_ctl: prevent removal of kernel controls
  sound: snd_ctl_remove_unlocked_id: simplify user control counting
  sound: snd_ctl_remove_unlocked_id: simplify error paths
  sound: snd_ctl_elem_add: fix value count check
2009-09-10 15:32:47 +02:00
Takashi Iwai
124e39b34d Merge branch 'topic/cs46xx' into for-linus
* topic/cs46xx:
  ALSA: cs46xx - Fix minimum period size
2009-09-10 15:32:46 +02:00
Takashi Iwai
9d2743f84d Merge branch 'topic/cmi8330' into for-linus
* topic/cmi8330:
  ALSA: cmi8330: Allow MPU-401-less operation
  ALSA: cmi8330: find OPL3 port automatically
  cmi8330: Add basic CMI8329 support
  ALSA: cmi8330: revert comments about AD1848 back
2009-09-10 15:32:45 +02:00
Takashi Iwai
d0064a1b22 Merge branch 'topic/cleanup' into for-linus
* topic/cleanup:
  ALSA: info - Use krealloc()
2009-09-10 15:32:43 +02:00
Takashi Iwai
b81e5ab34d Merge branch 'topic/azt3328' into for-linus
* topic/azt3328:
  ALSA: azt3328: fix previous breakage, improve suspend, cleanups
  ALSA: azt3328: large codec cleanup, add I2S port etc.
  ALSA: azt3328: fix Kconfig entry
2009-09-10 15:32:41 +02:00
Takashi Iwai
e0b3032bcd Merge branch 'topic/asoc' into for-linus
* 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
  ...
2009-09-10 15:32:40 +02:00
Takashi Iwai
45fae5c78d Merge branch 'topic/ali5451-cleanup' into for-linus
* topic/ali5451-cleanup:
  ALSA: ali5451: remove dead code
2009-09-10 15:32:38 +02:00
Mike Rapoport
2ba9fd0d15 [ARM] pxa: update pxa2xx-ac97.c to use 'struct dev_pm_ops'
Signed-off-by: Mike Rapoport <mike@compulab.co.il>
Signed-off-by: Eric Miao <eric.y.miao@gmail.com>
2009-09-10 19:15:37 +08:00
Manuel Lauss
cdc65fbe18 ASoC: au1x: PSC-AC97 bugfixes
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>
2009-09-08 19:21:27 +01:00
Takashi Iwai
b888d1ce82 ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128
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>
2009-09-08 18:15:17 +02:00
Takashi Iwai
9b151fec13 ALSA: dummy - Add debug proc file
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>
2009-09-08 14:46:49 +02:00
Takashi Iwai
4f7454a997 ALSA: Add const prefix to proc helper functions
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>
2009-09-08 14:45:06 +02:00
Takashi Iwai
6e5265ec34 ALSA: Re-export snd_pcm_format_name() function
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>
2009-09-08 14:26:51 +02:00
Takashi Iwai
33d7867458 ALSA: hda - Use auto model for HP laptops with ALC268 codec
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>
2009-09-08 11:07:56 +02:00
Sophie Hamilton
6148b130eb ALSA: cs46xx - Fix minimum period size
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>
2009-09-08 10:59:49 +02:00
Mark Brown
87831cb660 ASoC: Fix WM835x Out4 capture enumeration
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
2009-09-07 18:56:24 +01:00
Takashi Iwai
82a783f4bc ALSA: Remove struct snd_monitor_file from public sound/core.h
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>
2009-09-07 15:50:18 +02:00
Clemens Ladisch
f1bc07af9a sound: oxygen: work around MCE when changing volume
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>
2009-09-07 12:15:43 +02:00
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